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. Aprox. 10 of them are for our current Mitel PBX system. We also have two fax lines. Because of the proprietary nature of our Mitel phones, I will need to replace our current phones. I am looking at the Sipura SPA-841 phones as well as a few SPA-2000 ATAs for the few analog phones we have. My concern is for our incoming lines. I am not sure whether to go with a VOIP provider or to stay with our existing lines. A T-1 may also be an option for our phone lines. We could also use the t-1 for our internet. Does anyone know of a PSTN gateway that is fairly inexpensive that would allow us to use our existing phone lines? As I said earlier, the deciding factor is cost. I also want something that is reliable. Ideally, I would like to go with the option that is the least difficult to set up and maintain. I am the only IT person in our organization that knows how to work with Linux. I am basically looking for some guidance here. Has anyone dealt with a similar situation and how did you resolve it. Any help would be appreciated. -- Jason Fayre Colorado Center for the Blind
-----Original Message----- <snip> My concern is for our incoming lines. I am not sure whether to go with a VOIP provider or to stay with our existing lines. A T-1 may also be an option for our phone lines. We could also use the t-1 for our internet. Does anyone know of a PSTN gateway that is fairly inexpensive that would allow us to use our existing phone lines? As I said earlier, the deciding factor is cost. I also want something that is reliable. -Look into the cost of a T-1/PRI to replace your analog lines. This will be the easiest way to deal with your environment. One card in the * server (Digium single port T-1/PRI) with a single cable to plug in. You will also get additional features like DID (Direct Inward Dial) numbers. <snip> I am basically looking for some guidance here. Has anyone dealt with a similar situation and how did you resolve it. Any help would be appreciated. -you could use the existing analogs with a channel bank or some FXO gateways but those options in the end will cost more money. Jason Kawakami www.optellabs.com Salt Lake City, UT
jose luis campos wrote:> Hi All > > Finally, after some reading ;^), I can do some basic sip channel > configuration (two softphones communicating each other), the next step > i did was to register with a sip provider (voipjet), and with my free > 25cents I call my mother, my girlfriend and some friends (Yeahhh). > > What I notice is that my voice sounds strange, like with interference > (how i know this, I also spoke with my phone answer machine lol, hey > man Im excited!). The question is, how can I improve the quality of > the service, I now that if I rent a T1 I could made it, but all I got > is my own 512kbps connection provided by Mexican Monopoly (named > Telmex) provider, could someone please explain a roadmap to achieve > this goal, make asterisk work better. > > Thank you very much, any hel will be highly appreciated. > > Salu2Hola Jose, First thing I will mention is that you posted to the developer list. This is a topic for the user list. So I am replying on the user list and cc'ing you in case you haven't subscribed to that list. Second thing I am wondering is if you are running asterisk or just playing with softphones. If you are not running asterisk, you should be seeking help in the lists and forums for the softphones you are using. You need about 90kbps in and out to have a single high quality call in progress. You need to learn about codecs if you want to use less bandwidth(changes call quality). Be sure that you are not doing anything else that consumes all your bandwidth while evaluating call quality. Set you softphone to use 711u codec which is high bandwidth and high quality. If you still have problems it might be the telmex network causing it. Did you do a traceroute and ping to the voipjet server you are using for your calls? That would tell us something about your telmex connection quality.