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I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through).... Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd

I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a

Hi everyone - I have a nicely working system to which I'd like to add popups for incoming calls. Calls go into a queue, then all extensions ring. I'd like the agent that answers to call to get the popup on screen. I'm currently using Flash Operator Panel to get a popup (other suggestions welcome). Currently, all users get a popup when the call first goes into the queue

Hi - I've got a new install with a Sangoma A200 and a few GXP2000's. When users are talking over the Sangoma, they get a lot of sidetone (local echo). Internal calls are fine. Where do I adjust that? I assume its in zapata.conf somewhere? thanks Todd

Hi I've a Sangoma A200D with 2FXO and 2FXS. When using it with only the FXO module, it's all good. But when I put in the FXS module and connect the power, logs tells me not enough power. > Mar 31 14:11:54 phone kernel: [ 4761.246931] wanpipe1: Module 1: > Failed to powerup within 600 ms (8V : 72V)! > Mar 31 14:11:54 phone kernel: [ 4761.246937] wanpipe1: Module 1: Did

Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to

Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI> A No such command 'A' (type 'help' for help)

Hi everyone I just ordered a Sangoma A20001 with 2FXO ports - Does anyone have suggested reading pointers for what I'll need to do to get it working? I've only used VoIP in the past so don't know much about Sangoma drivers or Zaptel. I opted for the non-echo canceling card so I may need to do some tuning? Looking for reading... Hurl an URL at me! thanks! Todd

Hi - I am going to open port 5038 on my firewall so that I can use YAACID to spawn browser popups on an incoming call. My question is, under manager.conf, what are the suggested settings so that I can get the browser popups only? I'll be at different IPs so I can't lock it down that way.. I guess I don't need any write access? [managername] secret=secretword

Hi - Am I allowed to have multiple managers logged in with the same manager username at the same time? I'm referring to the id names in manager.conf. I expect so, but just want to check to help in troubleshooting a problem. thanks -todd-

Hi - I got a SPA3102. I've set it up without to many problems. If the unit looses power, calls to the PSTN are bridged which is nice. However, if the Asterisk server is unavailable (I turned it off to test), calls out are not bridged to the PSTN. I've rebooted the SPA3102 with the asterisk server off, but still it gives me no dial- tone. Under the configuration, Auto PSTN

Hi - I have a queue setup with 4 agents. The people working the queue tell me that when a new call goes into the queue, both the agent and the caller hear a tone. These are static agents - could it be ringing the line again, even though the agent is on a call? We are using GXP2000's with only 1 line programmed in. thanks Todd queue.conf > [601] > announce-frequency=0

Hi All- I have an asterisk server and GXP2000. If I want to send a call to someone else (external), I can transfer the call where I can announce it, and then send it over. But what I'd like is to start a 3-way conference, and then drop out. But if I do a conference button on the phone, and then drop out, the other two are not left to finish their conversation (the call is ended).

Hi - I guess it's not possible to use Asterisk BLF function for queues... Can someone confirm that? I'm looking for that type of function with calling queues. I have Grandstream gp2000 phones. thanks Todd

I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble.... thanks Todd

I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate

This might be a newbie question... I'm using a SIP trunk and trying to get DID line information on an incoming call. All I hear is a nice lady saying 'Zero' - then the call continues... Any suggestions? thanks Todd exten => s,n,Set(DIDID=(<${FROM_DID}>)) exten => s,n,SayNumber(DIDID) or exten => s,n,Set(FROM_DID=${EXTEN}) exten =>

I'm setting up an asterisk server where an administrator will not always be available in case of problems. While I expect problems to be rare, I need to be prepared. We're thinking of VoIP DID's and SIP phones so it's an all TCP/IP network. We could get a second server to substitute - What is involved in 'transferring' or 're- registering' the DID

Hi - I just got an A200 card with 1 FXO and 1 FXS module. Sadly, I can't make it work- currently, asterisk will not startup because of a bad module. Below are some log files/config files. If anyone has any suggestions, I'd appreciate it. I used Trixbox 2.0 and followed instructions on (http:// sangoma.editme.com/wanpipe-linux-asterisk-atHome) - no problems running through or