<div dir="ltr">Hi,<br><br>Eventually I did not went with 018, as they have the ugliest support I ever seen - they don't reply on emails. I asked twice from them to send oraat keva blank, and got nothing.<br>The broken "שלך" button on "צור קשר" page (under firefox) did not help either:<br>
<a href="http://www.018.co.il/contactus.asp">http://www.018.co.il/contactus.asp</a><br><br>Now I am looking in direction of 012 and have couple of questions to actual users of theirs voip service with Linux (if there are any):<br>
<a href="http://www.012.net.il/sales.aspx?docID=8639&FolderID=1005&lang=he&tabn=0">http://www.012.net.il/sales.aspx?docID=8639&FolderID=1005&lang=he&tabn=0</a><br><ul><li>Does the line management site (<a href="http://072web.net">http://072web.net</a>) works from firefox?</li>
<li>Do they provide SIP credentials, or are these hidden in the hw sip adapter they provide?</li><li>Can Asterisk be used with the service?<br></li></ul>Thank you in advance!<br><br>--<br clear="all">Arie<br><br>
<br><br><div class="gmail_quote">On Sun, Mar 8, 2009 at 03:01, e2xbegqsdyt21hfc <span dir="ltr"><<a href="mailto:e2xbegqsdyt21hfc@yahoo.com">e2xbegqsdyt21hfc@yahoo.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
I don't know the technical details or 018.<br>
I do know that 012 provides you with a sip adapter that is plugged between the bezeq/hot modem and your PC/ordinary phone. It could be that that adapter reserve some BW for the sip packets. In any case, my experience is that both 750K ADSl/cables is sufficient for a good quality phone service, with a lightly usage for internet and phone. The sip adapter I initially had was made in the US. I assume they use a US proven technology. My current sip adapter is made by AudioCodes. 012 claims that the sip adapter they provide should work with any ISP. I had two ISPs, and in my case their claim is justified.<br>
There is also a rumor that bezeq's new generation network uses sip over copper for the phone services.<br>
<br>
--- On Sat, 3/7/09, Arie Skliarouk <<a href="mailto:skliarie@gmail.com">skliarie@gmail.com</a>> wrote:<br>
<br>
> From: Arie Skliarouk <<a href="mailto:skliarie@gmail.com">skliarie@gmail.com</a>><br>
> Subject: Re: xfone 018 phone service and Linux<br>
> To: "linux-il" <<a href="mailto:linux-il@cs.huji.ac.il">linux-il@cs.huji.ac.il</a>><br>
> Date: Saturday, March 7, 2009, 11:21 PM<br>
<div><div></div><div class="h5">> Hi,<br>
><br>
> On Thu, Mar 5, 2009 at 13:54, Gilad Ben-Yossef<br>
> <<a href="mailto:gilad@codefidence.com">gilad@codefidence.com</a>>wrote:<br>
><br>
> > Arie Skliarouk wrote:<br>
> ><br>
> ><br>
> ><br>
> > How do they solve the latency problems inherent to any<br>
> internet connection?<br>
> ><br>
> > The round trip time to a well connected (read: 0%<br>
> packet loss) server in<br>
> > Israel from an Israeli ISP, where Israel here is<br>
> defined as "connected to<br>
> > the IIX", is under 50ms.<br>
> ><br>
><br>
> On a 150kbit upload ADSL upload of a 1500 bytes packet<br>
> takes about 85ms. To<br>
> send out the VoIP packet would take another 85ms (on<br>
> condition that you have<br>
> a really good QoS). This causes latency of 85-170ms with<br>
> jitter 85ms. The<br>
> VoIP speech (SIP protocol) has a lot of skips and is<br>
> unacceptable.<br>
><br>
> Theory aside, I would like to hear first-hand experiences<br>
> of people with 018<br>
> before I commit for a year of phone service with them.<br>
><br>
> --<br>
> Arie<br>
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<br>
<br>
<br>
</blockquote></div><br></div>