similar to: Finding good, objective reviews of major VoIP phones

Hi everyone, I'm new to the list, so I apologize if this is a trite question. We're an ISP with a barely used PRI, and also currently use asterisk with POTS service on an audiocodes gateway. We have a notion that we want to take on a new revenue stream, with our model being as such: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece

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

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hello All, I have been using the following phones with excellent success inside my lan: Cisco 7960 Polycom IP600 I have also been comfortable with the SPA-3000. I recently got a SPA-841 and the quality is aweful. Even in stock setup, it's like when I speak into the handset I sound like I am very far away. I want to get a hardphone for europe, but I need to be able to use a STUN server, and

Hi, I`m not actually asking for a comparaison between the two, I have one on hand and will make up my own mind. But I can't find much info on whether the Snom (370 to be exact) accepts FTP provisioning like the Polycom (but few others) do. Any Snom user can answer this one for me? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you.

In trying to get my services up and running, I've encountered the usual spate of first-time issues. I was wondering if there was a good FAQ or Howto on troubleshooting NAT issues. The equipment that I'm using is typically a Sipura ATA or hardphone (SPA-942) sitting behind either a Linux box as firewall (using IPtables and Arno's firewall) or else a Cisco IOS device running 12.4T

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

Hi, I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: me@gavinadams.org

Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a month ago and we're still waiting. Our account rep's communication with us on what the delay has been, has been terrible. Doug. -----Original Message----- From: Gavin Adams [mailto:me@gavinadams.org] Sent: Friday, January 27, 2006 8:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Hi, Are there any resellers of phones that can take power over ethernet in Australia? All I can find for sale online is the BT-10[12], which is cheap but not featureful enough, and the Snom 190, which is about right, but neither of them support PoE. I'm particularly intereseted in the Snom 220 with the keypad expansion for our receptionist. Although, could you make a PoE split-out cable

Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I

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

(found on dslr voip forum) http://www.freephoneproject.com/nexthop/ I think this is pretty cool. not too sure how stable/durable though, at least not until there are a lot more nodes providing service. also unsure of the legality vs. your average telco's consumer tos, but it's definitely not reselling. jeff pulver (crom bless him) seemed to be fairly amused, stuff like this fits

We need to resell systems that run commercial software on Linux. We were using Red Hat ES 3 originally. But we later realize that we cannot redistribute RHES without paying them huge subscription fee because of the trademark. I came across CentOS from WBEL mailing list. This looks like a godsend to me. I have some questions regarding CentOS. I would appreciate it if someone can provide me the

Hi! I am doing a lapply and for comparaison and I get that for is faster than lapply. What I have done: n<-100000 set.seed(123) x<-rnorm(n) y<-x+rnorm(n) rand.data<-data.frame(x,y) k<-100 samples<-split(sample(1:n),rep(1:k,length=n)) res<-list() t<-Sys.time() for(i in 1:100){ modelo<-lm(y~x,rand.data[-samples[[i]]])

Hello everybody This question has probably already been asked, but I'd like to have feedbacks about Yuxin hardphones Especially series 10, 100 and 200 ( and by the way i didnt found too much technical differences between those models). Is it better than budgettone, or so cheap hardphones ? I m' like everybody and dont want to waste money in bad phones :) So any experience is welcome.

None of the unlimited packages you get plan on you using them up. It's like a couple of years ago when the long distance carriers gave us the first phone call of the holiday for free, and people let the phone off hook for the entire week end. It's illegal for sure, and for a couple of reasons. a. you're not allow to resell anything someone else sold you and not pay taxes on it. b.

Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type: