similar to: Budgetone ringing volume

We have 2x BRI's connected to Asterisk which give us a total of 4 lines (using the bristuffed package). We would like to limit the number of incoming calls to 2 calls and if a 3rd call comes in, we would like this to go to another extension (voicemail or similar). Is this possible in Asterisk? Thanks -- James Doherty - Systems Engineer Zeald.com - Websites That Work! Web: www.zeald.com | Ph:

I'm using a Grandstream IP phone to call someone through our asterisk pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses 2x ZAP-HFC cards. When I call someone, if the call isn't answered and then I hang up, I get "487" coming up on the grandstream phone. If I pick up the receiver again and then hang up, the PBX starts calling me back and when I

Hi all, In our asterisk box we have a Fritz card and an HFC card. So far I've had the Fritz card working well, but its time to connect the HFC card up to the ISDN line and get it working. I've followed the instructions on voip-info.org here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install and here: http://www.voip-info.org/wiki-Asterisk+zaphfc+install But when it comes to

I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody. I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are pretty much the default, except in order to get the fcpci module to compile, I had to follow the instructions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install but the drivers would only compile if I left

I have a "Cologne Chip Designs GmbH ISDN network controller" in my asterisk box, and an ISDN line connected up to it. I downloaded the bri-stuff 0.1.0-RC4a package from junghanns.net and used that to compile asterisk using these instructions: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install When I try to call my asterisk box, my sip phone rings twice and then the

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. I've found a problem when I pickup a remote sip phone with *8. There're both budgetones 102 and are both in the same group. When one sip phone is ringing, I can pickup the call from another sip phone, but the first one keeps playing a loud busy signal... that don't go away until I receive another call or go off hook and then on hook on the first phone. I think that could be a

Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The

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

Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working?

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 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

Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the

Hi Folks, Since there isn't a grandstream forum AFAIK I guess someone here may be able to shed some light on this. Apologies if this is viewed as offtopic.. I think I may have killed the firmware on my Grandstream Budgetone 101. I found a source for the 1.0.5.30 firmware and made the files available over tftp. The phone downloaded the files but now doesn't boot and hangs with a blank

asterisk-users-request@lists.digium.com is believed to have said: >Budgetone and MWI > >The message button can be programmed to dial an extension that checks >voicemail >exten => 160,1,Voicemailmain(${CALLERIDNUM}) > Thanks, this is what I was thinking about. Still, how do you get the BT to dial 160? In my Asterisk setting I have the same mailbox numbers reused for the

I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian

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 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

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 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