similar to: VOIP adapters to connect PSTN lines to SIP phones

Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have an Asterisk 1.2.9.1 on a Debian Sarge distro connected to a VoIP provider via internet. I noticed Asterisk gets slow and behaves in strange manner if I unplug my internet cable from the PBX: for example I get incoming calls after seconds or I get no audio during calls. I thought it was something connected to DNS resolution so I put VoIP provider addresses inside /etc/hosts but

Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone ---- My own Asterisk ----- Outside lines But when it comes to smaller villages (I deal with people in tiny

Hi, I`ve been toying with an Aastra phone (9143i) wondering if it could be a good alternative to to the more expensive Polycom phones. One thing which I can't figure out, although it certainly looks simple, is to update the firmware though FTP (not TFTP). I have set the ftp provisioning server in the Aastra phone, and put the firmware file 9143i.st in the root folder where the

Hi, I'm stuck doing an install with Polycoms at a small office with no RJ-45. They went wireless 100%, poor them. I insist on using Polycom unless it's impossible because that's what I am standardized on for many reasons. What's the best way/device to turn a wired Polycom 501 (or any Polycom for that matter) into a WiFi phone? Mike -------------- next part --------------

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten

http://www.siliconvalley.com/mld/siliconvalley/business/international/as ia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and

Hello list, I am doing some work for a non profit group. As part of this, I am going to be putting in a 30 handset Asterisk solution. We are trying to keep the costs down as much as possible, as this job includes cabling, I am looking at POE solutions. On the switch side, I am considering something like some Netgear ProSafe FS726TP 24 port switches, or maybe the equivalent Linksys SRW224MP 24

Hello all, Our setup consists of an asterisk server (version 1.4.8 ) with a TDM400 card (4 FXO ports) and about 10 aastra 5Xi phones (2 phones are 57i and the rest 55i). The asterisk server is also configured with two SIP trunks. The problem that we're having is that if at any time there is an internet connection interruption to the asterisk server all the aastra phones freeze. If we disable

Hi, I have a few customers hosted on my Asterisk PBX. All incoming calls are coming from VoIP provider A and all outgoing calls are going to termination provider B (they happen to be different for now). On ONE of those customers I'd like to setup a channel limit, meaning they can have 4 calls max, either outgoing, incoming, or a combinaison of both. After that, I want it to ring busy.

Hello, I have a big old Fax machine that will only pick up a ring at 24volts and 20mA(minimum) ring[according to the technical specs manual]. None of my SIP -> Analog phone adapters supply this: Cisco - 50 volts SIPURA - 70 volts Handytone - who knows, but it doesn't work can anyone tell me if there are any analog adapters or channel banks that can send a 24V 20mA ring signal? Also,

Hello! I'd like to connect regular analog phones (Cordless Panasonic, for instance) to Asterisk. What's the cheapest FXS device on the market that can be used with Asterisk? Where can I find it? It has to be the cheapest to avoid huge import taxes, since I'm planning to buy in bulk. I'm also looking for BRI ISDN cards (cheap and used, preferable) that can be used in NT mode so I

Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

Hi, I know that if you have Polycom phones properly configured, you can use "sip notify polycom-check-cfg SIP_REGISTRATION_ID" to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike -------------- next part

Hey all, I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone

Folks, I am a newbie to the VOIP world and have a question (might as well sound silly to some). I would like to set up a PC-to-Phone call from my desktop to a regular PSTN number. Does the Asterisk PBX itself act as a VOIP-PSTN gateway or do I have to subscribe to a VOIP provider for this? Are there any free IP-PSTN gateways which I can subscribe to? Do I need any specific hardware to set up such

Hello, I'm looking to connect Asterisk with three (four in the future) PSTN lines, and would like to get some opinions on the TDM400 Digium card, vs. sip gateways like the Mediatrix 1204, vs. other hardware solutions I'm not yet aware of. I need the ability to prioritize which PSTN lines are used for outgoing calls (I understand this can be done with the Mediatrix --

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

Hi list, would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side,