similar to: Beginers Help - Hardware selection

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? I built another route 01144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest

BR C.

I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

Hello All, Just looking some comments from gurus about this proprietary systems and phones: Inter-Tel Eclipse2 Model name: IP PhonePlus I did not find anything useful or reasonable about their products on their website or even in Internet.... except sales. -- Thanks and regards, Vasyl Rublyov

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers

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

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

Hello Everyone, I am looking into using Asterisk as our company PBX and voicemail system. I am very familiar with Linux, but the VOIP stuff is new for me. We are a non-proffit organization, so keeping things as cheap as possible is very important. I am looking on some advice for best implementing Asterisk. Here is a rundown of our current system: We have 14 phone lines coming into the building.

Hey all, Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

Hello list, I am new to Asterisk and have a few questions, I hope you can tell me if Asterisk can do what I need. I'd like to create a simple IVR menu that will allow people to enter prices using their telephone. The goal of my project is to create a central database of gasoline prices in my area, which will be published online. I would like to have an automated system place a call

G'Day All, Newbie here. How can I go about troubleshooting a fast busy when I dial my the phone number on my * server? Thanks. Ferg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041012/a669d795/attachment.htm

I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... Dustin Moore

Guys. I just downloaded the recent firmware for GS GXP 2000 and I must say the phone works great but... How do you make the conf button work?????? Anybody done that? Also, with great dissapointment I must ask, where is ILBC support? GS web page mentions it and the manual says it supports it almost using bolds :) soooooo where is it???? Any light on this?