similar to: The Budgetone 100

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

Hi All I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S based ISDN card using ISDN4Linux. I have setup the BudgeTone-100's to use Early Dial which for calling between the three phones works well, but for the external calls using the following extension exten => _9.,1,Dial(Modem/g1:${EXTEN:1}) Only the first digit is dial on the ISDN Line. Does anyone know of

I have two problems with Asterisk and my Budgetone 100s. 1. Sometimes when outside callers dial my Budgetone it will ring twice, then the caller will receive a busy signal. This happens intermittently and I cannot pin down the circumstances of when it works and when it doesn't. Asterisk logging doesn't give me too much info when this happens: *CLI> -- Executing

Hello, I have been working on getting Asterisk up and running with a couple Grandstream hard phones (Budgetone 100s) and a couple softphones. Things work pretty well but we have discovered an odd problem. A calls B and then A can put the call on hold and pick it back up. A calls B and B can put the call on hold but cannot pick it back up. These tests are internal on the hard phones. The

Has anyone seen/dealt with the Grandstream BudgeTone-100 producing double dialing with asterisk? Thanks JDT Example: (password input = 101) -- Playing 'vm-password' -- Incorrect password '11001' for user '101' (context = <any>) -- Playing 'vm-incorrect' -- Playing 'vm-password' -- Incorrect password '110011' for user

I have a strange problem in that I have put a budgetone out on the internet that connects to my * server that's behind a firewall. They can call me I can call them and it works fine. However, I have setup a link to sipdiscount on my * server. If the budgetone user calls via my * box to sipdiscount all the budgetone user hears is silence and the called person hears silence as well when they

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

Hello all, Does anyone know how to successfully transfer a call using the GrandStream Budgetone 100 phones? I've read multiple posts talking about hitting flash, the dialing, then flash again, etc. Some posts talk about using the transfer button, then dial, then flash. Anyways, it seems that I am able to put the caller on hold (whereas they hear hold music) by pressing flash or

Hi, I'm considering giving the Grandstream BudgeTone-102 phones a try. I've been using Cisco 7960's to date, but the low cost of the Grandstream phones are hard to ignore. I have two questions: 1) Does the message waiting indicator on the BudgeTone's work with Asterisk? 2) The one line 12-digit LDC concerns me a bit. Is the LCD able to display both the CID number and name on

Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know

Hello to all. I have a couple of budgetones connected to Asterisk server. I can establish calls using budgetone with no problem, but when I hang up a Budgetone, Asterisk does not detect the hangup. It seems that the communication goes on in spite of budgetone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet

The process for transfering a call with the Bugetone is as follows.. 1. Press "transfer", you will get a dial tone.. 2. Dial in the extension to wish to transfer to.. 3. Press the "Redial".. (on the newer phones this is the "send" button) You don't need the "t" option on your dial string to do transfers with a budgetone.. Later.. > With the right

I noticed that the BudgeTone (I have the 102) with the latest firmware tries to download a file called cfg.txt (presumably the configuration) and a file called ring.bin (presumably a ringer) from the tftp server. The ring-in sound on the budgetone is the same as the ring-out sound and that is going to be confusing for users. I contacted GrandStream and was informed that both of these formats are

Hi, Just playing with a couple of Budgetone 102 phones and they are pretty good for the price. The only problem i'm having at the moment is when I get a voicemail on the Asterisk box the LCD flashes. Dialing *98 goes to the VoiceMail Manager, and asks for mailbox, I enter 201, then asks for password, enter my voicemail password set in the Extensions -> webadmin, then hit the

Hi I am setting up an Asterisk System in an office environment, Incoming and Outgoing calls are working ok, but i am having a few strange problems regarding call waiting. With the SNOM 200 (firmware 2.02t) phones, if you are on a call and a 2nd call comes in, the call waiting beep is played and the light flashes, but if you hang up the 1st call, instead of the phone ringing, it connects the

Hi everyone. I see many posts complaining about Grandstream equipment, so I thought I tell the other side of the story as well. We use Grandstream Budgetone 101 phones in our office and they work extremely well. We have no echo, no crashes, no sudden resets, they "just work". We provision them using TFTP. Our asterisk box has 2 HFCPCI ISDN BRI cards. It's running asterisk 1.2.0 on

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,