<?xml version="1.0" encoding="UTF-8"?>
<rdf:RDF xmlns:rdf="http://www.w3.org/1999/02/22-rdf-syntax-ns#" xmlns="http://purl.org/rss/1.0/" xmlns:taxo="http://purl.org/rss/1.0/modules/taxonomy/" xmlns:dc="http://purl.org/dc/elements/1.1/" xmlns:syn="http://purl.org/rss/1.0/modules/syndication/" xmlns:admin="http://webns.net/mvcb/">
  <channel rdf:about="http://blog.gmane.org/gmane.comp.telephony.pbx.asterisk.biz">
    <title>gmane.comp.telephony.pbx.asterisk.biz</title>
    <link>http://blog.gmane.org/gmane.comp.telephony.pbx.asterisk.biz</link>
    <description/>
    <syn:updatePeriod>hourly</syn:updatePeriod>
    <syn:updateFrequency>1</syn:updateFrequency>
    <syn:updateBase>1901-01-01T00:00+00:00</syn:updateBase>
    <items>
      <rdf:Seq>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32915"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32914"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32913"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32912"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32911"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32910"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32909"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32908"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32907"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32906"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32905"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32904"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32903"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32902"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32901"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32900"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32899"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32898"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32897"/>
        <rdf:li rdf:resource="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32896"/>
      </rdf:Seq>
    </items>
    <image rdf:resource="http://gmane.org/img/gmane-25t.png"/>
    <textinput rdf:resource=""/>
  </channel>
  <image rdf:about="http://gmane.org/img/gmane-25t.png">
    <title>Gmane</title>
    <url>http://gmane.org/img/gmane-25t.png</url>
    <link>http://gmane.org</link>
  </image>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32915">
    <title>Planned service outage for community services</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32915</link>
    <description>&lt;pre&gt;On May 31, 2012 from approximately 9:00AM to 12:00PM (Central Daylight 
Time, GMT-5), the servers that Digium uses to provide many services to 
the Asterisk community will be relocated. This will mean that these 
services will be unavailable during most, if not all, of this time 
window. Once the move is complete, the services will be available again, 
with no user-visible changes.

The services affected include:

bamboo.asterisk.org
code.asterisk.org
downloads.digium.com
downloads.asterisk.org
git.asterisk.org
issues.asterisk.org
packages.asterisk.org
reviewboard.asterisk.org
svn.asterisk.org
svnview.digium.com
wiki.asterisk.org

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Asterisk Development Team</dc:creator>
    <dc:date>2012-05-23T14:45:00</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32914">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32914</link>
    <description>&lt;pre&gt;
23 maj 2012 kl. 14:35 skrev Alex Balashov:


Absolutely. But it's at least an estimate better than 500 ms in most situations. It does affect the quality of the call.

/O
--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Olle E. Johansson</dc:creator>
    <dc:date>2012-05-23T12:45:49</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32913">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32913</link>
    <description>&lt;pre&gt;That comes down to whether userspace, SIP stack-level OPTIONS pings are a "good estimate of RTT". :-)

--
This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On May 23, 2012, at 5:28 PM, "Olle E. Johansson" &amp;lt;oej&amp;lt; at &amp;gt;edvina.net&amp;gt; wrote:


--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Alex Balashov</dc:creator>
    <dc:date>2012-05-23T12:35:39</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32912">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32912</link>
    <description>&lt;pre&gt;Just for the archives:

Don't forget that you can use the SIPPEER() dialplan funciton to check the status of the peer with qualify=on before you place the call in the dialplan.

/O
--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Olle E. Johansson</dc:creator>
    <dc:date>2012-05-23T12:34:58</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32911">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32911</link>
    <description>&lt;pre&gt;
23 maj 2012 kl. 14:21 skrev Kevin P. Fleming:


Form the RFC:

"T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms."

"The default value for T1 is 500 ms. T1 is an estimate of the RTT between the client and server transactions.
Elements MAY (though it is NOT RECOMMENDED ) use smaller values of T1 within closed, private networks
that do not permit general Internet connection. T1 MAY be chosen larger, and this is RECOMMENDED if it
is known in advance (such as on high latency access links) that the RTT is larger. Whatever the value of T1,
the exponential backoffs on retransmissions described in this section MUST be used."

I can't see how this is not RFC 3261 compliant. We have a good estimate of the RTT and use it.

/O
--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Olle E. Johansson</dc:creator>
    <dc:date>2012-05-23T12:28:41</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32910">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32910</link>
    <description>&lt;pre&gt;Ah, I see.  Yes, I embrace the pragmatism of such a design decision, though I do think that a little more emphasis could be put on the fact that rollover is not based on standard T1 values.

--
This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On May 23, 2012, at 5:21 PM, "Kevin P. Fleming" &amp;lt;kpfleming&amp;lt; at &amp;gt;digium.com&amp;gt; wrote:


--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Alex Balashov</dc:creator>
    <dc:date>2012-05-23T12:25:57</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32909">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32909</link>
    <description>&lt;pre&gt;
Neither. If 'qualify' is enabled for a peer, the T1 timer for that peer 
is reduced to the average of its response time to the OPTIONS pings that 
'qualify' generates (with a default minimum of 100ms). This behavior can 
be overridden by changing the minimum T1 timer value as I posted previously.

While this behavior is technically not RFC3261 compliant (and I've had 
discussions about it with at least one of the RFC's authors), it's quite 
useful in making decisions about whether a peer has become unavailable 
more quickly than would normally be possible. For a local peer that 
responds to OPTIONS requests in 100ms or less, if that peer stops 
responding, Asterisk will be able to make that determination in 
approximately 6 seconds, instead of the 32 seconds that would normally 
be required.

&lt;/pre&gt;</description>
    <dc:creator>Kevin P. Fleming</dc:creator>
    <dc:date>2012-05-23T12:21:54</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32908">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32908</link>
    <description>&lt;pre&gt;You mean Asterisk uses a Timer T1 value of 250 by default? Or just for OPTIONS requests?

--
This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On May 23, 2012, at 5:06 PM, "Kevin P. Fleming" &amp;lt;kpfleming&amp;lt; at &amp;gt;digium.com&amp;gt; wrote:


--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Alex Balashov</dc:creator>
    <dc:date>2012-05-23T12:09:20</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32907">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32907</link>
    <description>&lt;pre&gt;

Setting 't1min=500' will override this behavior of the 'qualify' 
mechanism, and then it will do exactly what you want.

&lt;/pre&gt;</description>
    <dc:creator>Kevin P. Fleming</dc:creator>
    <dc:date>2012-05-23T12:06:52</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32906">
    <title>Re: Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32906</link>
    <description>&lt;pre&gt;
Not sure but isn't it possible to set the time for a qualify with the
qualify and qualifyfreq config options?

Regards,
Partick

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Patrick Lists</dc:creator>
    <dc:date>2012-05-23T06:28:19</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32905">
    <title>Quote for feature: Check to see if a peer is up</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32905</link>
    <description>&lt;pre&gt;Hi,

 

We want an option for sip.conf that would be similar to qualify. We need to
know if the peer is responding to OPTIONS packets or not. This way if it is
down we know right away to continue in the dial plan. The issue with qualify
is that if the response time is 250 MS and when Asterisk sends an invite it
does not get a response in 250 MS then it sends the invite again which then
"irritates" some gateways.

 

Regards,

 

Dovid

 

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Dovid Bender</dc:creator>
    <dc:date>2012-05-23T06:14:28</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32904">
    <title>Re: RFP: GSM &lt;-&gt; VoIP Call-Center across 4 Countriesin Asia</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32904</link>
    <description>&lt;pre&gt;p.s. Forgot one requirement: a simple web-interface through which to listen to the call recordings…

On 2012-05-23, at 1:31 AM, Jonathan Barratt wrote:



--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Jonathan Barratt</dc:creator>
    <dc:date>2012-05-22T18:34:43</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32903">
    <title>RFP: GSM &lt;-&gt; VoIP Call-Center across 4 Countries inAsia</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32903</link>
    <description>&lt;pre&gt;Hi All,

For any who might be interested in this sort of work, we have a project coming up in the near future for which we will require a call-center in one country and trunk-scale connections between it and two and later three other countries, all in Asia. The end users will call into GSM SIMs hosted by Asterisk servers situated in their native country, and then SIP will handle the international relay between those hubs and the call-center staff at our primary location; who themselves will also have GSM SIMs to receive direct calls from domestic clients.

The call center's dialplan will need to be arranged into five different groups to represent the five different companies that will all be working under the same umbrella. The current numbers in terms of SIM Cards/GSM Channels &amp;amp; Day Operators &amp;amp; Night Operators are: 
18 &amp;amp; 15 &amp;amp; 6,
14 &amp;amp; 8 &amp;amp; 3,
34 &amp;amp; 4 &amp;amp; 1,
21 &amp;amp; 7 &amp;amp; 3, 
4 &amp;amp; 3 &amp;amp; 1. 

So a total of about 91 channels and 51 operators to start, with rapid growth expected for at least two years after the first 6 months. There are three different mobile network operators serving the call-center site, and a similar variety in numbers at the first three other countries.

We need all the usual bells and whistles: ACD, Call Recording, Queue Monitoring &amp;amp; Statistics, ability to pause out of queue for short breaks, Agent Metrics (very important), MoH, blind transfers, but not voicemail or call parking. Agents will use soft-phones to start but we're prepared to move to real phones if quality demands it. No need for screen-pops or desktop computer integration yet. One other big item: QoS on the LAN —  we have none as of yet, are currently using a Debian box as our router and unmanaged gigabit switches, so we'll definitely need this added.

I don't need a formal proposal yet, if you're interested then please just send your qualifications and a rough ball-park estimate on time and *labor* costs but don't stress on them, they're not something you'd ever be held to: I will narrow the pool of applicants down to a short-list and then ask for more detailed, realistic figures as part of what would be considered an official proposal.

Thanks in advance!
Jonathan
CTO
IntelligentMillionaire.com
--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Jonathan Barratt</dc:creator>
    <dc:date>2012-05-22T18:31:52</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32902">
    <title>have a try</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32902</link>
    <description>&lt;pre&gt;i ordered an iphone4s and mac from this eshop , 20% Off all orders in May.

now i had recive it , i like it very much

so i tell you , hope you can try too

take a look :*&amp;lt;depthdeals.com&amp;gt;*

regards


&lt;/pre&gt;</description>
    <dc:creator>Alexander Argov</dc:creator>
    <dc:date>2012-05-20T02:57:44</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32901">
    <title>Independent RespOrg</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32901</link>
    <description>&lt;pre&gt;I am interesting in hearing about experiences with Independent RespOrgs

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000



 

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Don Kelly</dc:creator>
    <dc:date>2012-05-17T16:04:27</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32900">
    <title>Re: Looking for Israel "Kosher" DID</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32900</link>
    <description>&lt;pre&gt;


----------------------------------------------------------------------

Message: 1
Date: Mon, 14 May 2012 11:27:49 -0700 (PDT)
From: Steve Edwards &amp;lt;asterisk.org&amp;lt; at &amp;gt;sedwards.com&amp;gt;
Subject: Re: [asterisk-biz] Looking for Israel "Kosher" DID
To: Commercial and Business-Oriented Asterisk Discussion
&amp;lt;asterisk-biz&amp;lt; at &amp;gt;lists.digium.com&amp;gt;
Message-ID:
&amp;lt;alpine.DEB.2.00.1205141125020.16899&amp;lt; at &amp;gt;localhost.localdomain&amp;gt;
Content-Type: TEXT/PLAIN; format=flowed; charset=US-ASCII



On Mon, 14 May 2012, C F wrote:


My fail. I should have googled first. I did after Avi's response and 
learned there are also Islamic phones.

Amazing what you can learn when you look for yourself.

&lt;/pre&gt;</description>
    <dc:creator>Dovid Bender</dc:creator>
    <dc:date>2012-05-15T21:43:17</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32899">
    <title>Re: Looking for Israel "Kosher" DID</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32899</link>
    <description>&lt;pre&gt;

On Mon, 14 May 2012, C F wrote:


My fail. I should have googled first. I did after Avi's response and 
learned there are also Islamic phones.

Amazing what you can learn when you look for yourself.

&lt;/pre&gt;</description>
    <dc:creator>Steve Edwards</dc:creator>
    <dc:date>2012-05-14T18:27:49</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32898">
    <title>Re: Looking for Israel "Kosher" DID</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32898</link>
    <description>&lt;pre&gt;Here is first the gooooooooogle answer:

http://www.google.com/search?q=rabbi+ban+cell+phones+internet&amp;amp;ie=UTF-8&amp;amp;oe=UTF-8&amp;amp;hl=en&amp;amp;client=safari#sclient=tablet-gws&amp;amp;hl=en&amp;amp;client=safari&amp;amp;tbo=d&amp;amp;q=rabbi+ban+internet&amp;amp;oq=rabbi+ban+internet&amp;amp;aq=f&amp;amp;aqi=&amp;amp;aql=1&amp;amp;gs_l=tablet-gws.3...69073.71340.0.72126.12.12.0.0.0.1.199.1367.7j5.12.0.crf1.1.0.0.0TJaP4R0fWs&amp;amp;pbx=1&amp;amp;bav=on.2,or.r_gc.r_pw
.,cf.osb&amp;amp;fp=6dc489c4b06a532f&amp;amp;biw=1024&amp;amp;bih=672

Basically social media, texting, blogs, porn is bad for the religious soul.
In israel they came up with phones that don't have Internet and text
capabilities. But in order to enforce it they came up with specific
exchanges that identify the number as a kosher phone. Therefore congregants
that don't have a kosher number could be called up by their rabbi how come
they don't have one.

In the case of the oven (btw, you'll be surprised how many home appliances
have this observant feature) a religious Jew is prohibited to change temps
on a oven on sabbath as well as prohibited from putting anything into a hot
oven. So if one wants hot food 14 hours into the sabbath their only option
if they are using an oven is to have the oven run non stop for the entire
sabbath. Because most modern ovens have a safety feature of turning the
oven off after 12 hours or so they have the observant feature to bypass
that.




On Monday, May 14, 2012, Steve Edwards wrote:

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>C F</dc:creator>
    <dc:date>2012-05-14T16:14:51</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32897">
    <title>Re: Looking for Israel "Kosher" DID</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32897</link>
    <description>&lt;pre&gt;I don't see how it quite works out (some can be circumvented, some ends up
costing people extra money), I just know the ramifications of it:

You can get "kosher" mobile phones in Israel, with service tied to it, that
can't have SMS, Internet, and the phones don't have cameras, etc. There are
identified with a special prefix at different carriers.

On to the economics:
After it's was out for a few years (about 2.5-3yrs ago), it became the
absolute cheapest way to call between mobiles. Most mobile phones charge
8-13cents/minute for calls -- yet the kosher phones started to come with
300, 500, or even 1000 "free" minutes to other kosher mobiles, even at a
different carrier.
It's only in the last.. 6 months or so that other deals for regular mobiles
to call other mobiles at a reasonable price came about. (as wholesale term
to Israel mobile has dropped under 4cents).

Anyway - so to use a calling card, the cell phone user needs to have free
minutes to be able to call the access number or it's highly unlikely that
it's worth it (because there are other ways of dialing for 5cents/minute or
less to USA.. which I help with too.).
Hence, there have been for years calling cards with a kosher access number.
They occasionally get noticed and are put on the blocked list meaning of
"non-kosher" numbers.
I've seen many phone deals start to gravitate back towards only free kosher
minutes, and not free landline minutes (a trend in the last year) so that
makes the normal landline calling card access number not able to be used by
everyone. (If you need an Israel landline number, be in touch.)

-Avi Marcus
BestFone


On Mon, May 14, 2012 at 7:25 AM, Steve Edwards &amp;lt;asterisk.org&amp;lt; at &amp;gt;sedwards.com&amp;gt;wrote:

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Avi Marcus</dc:creator>
    <dc:date>2012-05-14T05:55:19</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32896">
    <title>Re: Looking for Israel "Kosher" DID</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32896</link>
    <description>&lt;pre&gt;

I have to admit my ignorance and curiosity. (I was 'stunned' when I 
discovered that my oven was 'observant.')

Can you explain what is a Kosher number?

&lt;/pre&gt;</description>
    <dc:creator>Steve Edwards</dc:creator>
    <dc:date>2012-05-14T04:25:14</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32895">
    <title>Re: Looking for Israel "Kosher" DID</title>
    <link>http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32895</link>
    <description>&lt;pre&gt;I thought they all route to rabbinic switchboard operators.

--
Alex Balashov - Principal 
Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Decatur, GA 30030 
Tel: +1-678-954-0670 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

C F &amp;lt;shmaltz&amp;lt; at &amp;gt;gmail.com&amp;gt; wrote:

--
_____________________________________________________________________
&lt;/pre&gt;</description>
    <dc:creator>Alex Balashov</dc:creator>
    <dc:date>2012-05-14T00:13:03</dc:date>
  </item>
  <textinput rdf:about="http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.asterisk.biz">
    <title>Search Engine</title>
    <description>Search the mailing list at Gmane</description>
    <name>query</name>
    <link>http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.asterisk.biz</link>
  </textinput>
</rdf:RDF>

