search for: lightwavetech

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the * voicemail. I can't seem to get it to timeout regardless of how many seconds I set it to. I assume this has something to do with the fact that * considers the call answered as soon as t...

Hi, I am using a Netjet-s ISDN Card, and am having some trouble dialling out (Incoming Works Fine). TRUNK=Modem/ttyI0 exten => _90XXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _90XXXXXXXXX,2,Congestion I get the following when diallingout: -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Modem/ttyI0/04XXXXXXXX||Ttm") in new

I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the

> -----Original Message----- > From: Dale [mailto:dale-list@lightwavetech.com] > Sent: Tuesday, January 18, 2005 1:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] QoS tagging - can Asterisk do this, > if not,what do you recommend? > > So my question is, does Asterisk offer the > ability to mark the...

Here is the situation: A T1 router going into an office which then plugs into the firewall box then into the switch. None of these devices support QOS.. Is there some sort of box/device that I can place between the T1 router and the firewall box which will allow me to prioritize voice traffic on this link? I can't change the T1 router to something that supports QOS because it has

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

It seems that all my CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. Any help would be appreciated. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

What does "t" mean in a CDR entry? This is in place of where the number that was dialed normally goes. For one IAX termination provider it always has a t instead of the number dialed. Also, we always see the word "hunguup" in the same record entry. This is the provider we have set to our secondary not primary. Is it transfer of some sort? I don't think there was a

After getting zaptel from the CVS server, compiling and installing it I type: modprobe zaptel and all is well. Then I type: modprobe wctdm and I get this: modprobe: Can't locate module wctdm Any idea why? I did this yesterday but with the CVS head of Asterisk and I got by this part without a problem. I reinstalled it all today because I wanted the stable release on our server since we

After installing the stable version of * and the Zaptel drivers with a TDM400 card using 1 FXO module on port 4, I start Asterisk and get this rolling up my screen thousands of times: Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while

I have my main * box setup for all incoming and outgoing calls to and from our SPA-2000's. I have now setup another * box in a different location and I would like the SPA's to send all outgoing calls out through the new * server but continue registering with the old * server so all incoming calls will still be routed through the old server to our SPA's. In the SPA-2000 config

Can someone point me to some docs that explain this or give me a direction to go in. I have seen docs on this in the past but can't seem to dig em up now when I need them. Basically I want one Asterisk server to be the traffic cop and send some calls directly to ATA's and some calls to another Asterisk server, the other Asterisk server will then direct the calls to the end users ATA

I know this may sound like a stupid question but I will put on my flame retardant suit and ask anyway. Is there any way to use/allow SIP reinvite and still track the length of the call? I realize that the whole idea of reinvite is that it takes the proxy out of the media path which, from what I understand also kills the proxy's ability to track the start/end time of the call for billing

We are in the process of ordering a Voice PRI to plug into Asterisk. Of course we will be buying a card from Digium for this. Question is this, there seem to be MANY options technically when ordering this PRI (in the US) but since this is the first time ordering a voice circuit I am clueless as to what options we need. Any clues would be helpful or maybe something has already been written

Hello, I am trying to build up a pretty meaty Asterisk box after doing our initial testing and playing on a 1ghz system. Right now I have decided on a prebuilt system which I normally don't do but thought it seemed like a good deal. I have included the initial specs below, I will be adding another 1 GB of RAM for a total of 2 GB. My first question is regarding the serial ATA drives... I

Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com

We are looking for a good firewall replacement which will basically do pot blocking and QOS. Our current solution just plain stinks.. We basically need to handle the traffic of a few web servers, mail server and asterisk box. The most traffic this device will need to handle is what can be shoved through a T1. I don't mind buying an appliance to get something solid but IP Cop just looks

We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in