http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz hourly 1 1901-01-01T00:00+00:00 http://gmane.org/img/gmane-25t.png http://gmane.org http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32919 <pre>Congratulations And Welcome Rusty! well be happy to colaborate ;) ----- Original Message ----- From: "Kevin P. Fleming" &lt;kpfleming&lt; at &gt;digium.com&gt; To: "Commercial and Business-Oriented Asterisk Discussion" &lt;asterisk-biz&lt; at &gt;lists.digium.com&gt; Sent: Friday, May 25, 2012 5:41 PM Subject: [asterisk-biz] Digium's new Community Support Manager - Rusty Newton __________ Information from ESET NOD32 Antivirus, version of virus signature database 6830 (20120126) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -- _____________________________________________________________________ </pre> Josef Grand 2012-05-25T17:35:26 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32918 <pre> 25 maj 2012 kl. 16:41 skrev Kevin P. Fleming: Welcome Rusty! /O -- _____________________________________________________________________ </pre> Olle E. Johansson 2012-05-25T14:56:18 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32917 <pre> Congratulations Rusty! Alistair Cunningham +1 888 468 3111 +44 20 799 39 799 http://integrics.com/ -- _____________________________________________________________________ </pre> Alistair Cunningham 2012-05-25T14:45:53 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32916 <pre>We'd like you all to help us welcome Rusty Newton to Digium's Asterisk development and community support team! Rusty has been with Digium for over five years, starting in the Technical Support department and then moving to a sales position where he assisted customers with Asterisk and Switchvox solutions to their business needs. Prior to joining Digium he spent more than five years in the telecom industry, installing, configuring and maintaining PBXs. A couple of weeks ago he moved into a new role (for him and for Digium), Community Support Manager. In this role he'll be the primary person responsible for ensuring that Digium's community services are providing what the community members need, that the systems are operating properly, and that issues and questions are getting the attention they deserve. He'll be working closely with our Community Director as well, especially for events like AstriCon and others. He works directly with the software development team at Digium, which will allow him to focus almos</pre> Kevin P. Fleming 2012-05-25T14:41:12 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32915 <pre>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 -- _____________________________________________________________________ </pre> Asterisk Development Team 2012-05-23T14:45:00 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32914 <pre> 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 -- _____________________________________________________________________ </pre> Olle E. Johansson 2012-05-23T12:45:49 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32913 <pre>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" &lt;oej&lt; at &gt;edvina.net&gt; wrote: -- _____________________________________________________________________ </pre> Alex Balashov 2012-05-23T12:35:39 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32912 <pre>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 -- _____________________________________________________________________ </pre> Olle E. Johansson 2012-05-23T12:34:58 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32911 <pre> 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 -- _____________________________________________________________________ </pre> Olle E. Johansson 2012-05-23T12:28:41 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32910 <pre>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" &lt;kpfleming&lt; at &gt;digium.com&gt; wrote: -- _____________________________________________________________________ </pre> Alex Balashov 2012-05-23T12:25:57 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32909 <pre> 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. </pre> Kevin P. Fleming 2012-05-23T12:21:54 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32908 <pre>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" &lt;kpfleming&lt; at &gt;digium.com&gt; wrote: -- _____________________________________________________________________ </pre> Alex Balashov 2012-05-23T12:09:20 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32907 <pre> Setting 't1min=500' will override this behavior of the 'qualify' mechanism, and then it will do exactly what you want. </pre> Kevin P. Fleming 2012-05-23T12:06:52 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32906 <pre> Not sure but isn't it possible to set the time for a qualify with the qualify and qualifyfreq config options? Regards, Partick -- _____________________________________________________________________ </pre> Patrick Lists 2012-05-23T06:28:19 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32905 <pre>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 -- _____________________________________________________________________ </pre> Dovid Bender 2012-05-23T06:14:28 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32904 <pre>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: -- _____________________________________________________________________ </pre> Jonathan Barratt 2012-05-22T18:34:43 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32903 <pre>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; Day Operators &amp; Night Operators are: 18 &amp; 15 &amp; 6, 14 &amp; 8 &amp; 3, 34 &amp; 4 &amp; 1, 21 &amp; 7 &amp; 3, 4 &amp; 3 &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 mont</pre> Jonathan Barratt 2012-05-22T18:31:52 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32902 <pre>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 :*&lt;depthdeals.com&gt;* regards </pre> Alexander Argov 2012-05-20T02:57:44 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32901 <pre>I am interesting in hearing about experiences with Independent RespOrgs --Don Don Kelly PCF Corp People Come First 651 842-1000 -- _____________________________________________________________________ </pre> Don Kelly 2012-05-17T16:04:27 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32900 <pre> ---------------------------------------------------------------------- Message: 1 Date: Mon, 14 May 2012 11:27:49 -0700 (PDT) From: Steve Edwards &lt;asterisk.org&lt; at &gt;sedwards.com&gt; Subject: Re: [asterisk-biz] Looking for Israel "Kosher" DID To: Commercial and Business-Oriented Asterisk Discussion &lt;asterisk-biz&lt; at &gt;lists.digium.com&gt; Message-ID: &lt;alpine.DEB.2.00.1205141125020.16899&lt; at &gt;localhost.localdomain&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. </pre> Dovid Bender 2012-05-15T21:43:17 http://permalink.gmane.org/gmane.comp.telephony.pbx.asterisk.biz/32899 <pre> 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. </pre> Steve Edwards 2012-05-14T18:27:49 Search the mailing list at Gmane query http://search.gmane.org/?group=$group=gmane.comp.telephony.pbx.asterisk.biz