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	<title>Asterisk FAQs</title>
	<atom:link href="http://asteriskfaqs.org/feed" rel="self" type="application/rss+xml" />
	<link>http://asteriskfaqs.org</link>
	<description>... Asterisk Open Source PBX Frequent Asked Questions</description>
	<lastBuildDate>Sat, 19 May 2012 08:03:02 +0000</lastBuildDate>
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		<item>
		<title>Extensions routing</title>
		<link>http://asteriskfaqs.org/2012/05/19/asterisk-users/extensions-routing.html</link>
		<comments>http://asteriskfaqs.org/2012/05/19/asterisk-users/extensions-routing.html#comments</comments>
		<pubDate>Sat, 19 May 2012 07:48:07 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[extension]]></category>
		<category><![CDATA[Greetings]]></category>
		<category><![CDATA[load balancing]]></category>
		<category><![CDATA[server1]]></category>
		<category><![CDATA[server2]]></category>
		<category><![CDATA[sip]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32086</guid>
		<description><![CDATA[Greetings! I&#8217;ve been playing around with &#8220;clustering&#8221; some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions between them? I mean, if today user with extension 101 is [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/19/asterisk-users/extensions-routing.html/feed</wfw:commentRss>
		<slash:comments>1</slash:comments>
		</item>
		<item>
		<title>BroadVoice Unlimited World PLUS &#8211; Dialplan Update (18/May/2012)</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/broadvoice-unlimited-world-plus-dialplan-update-18may2012.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/broadvoice-unlimited-world-plus-dialplan-update-18may2012.html#comments</comments>
		<pubDate>Sat, 19 May 2012 01:48:41 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[BroadVoice]]></category>
		<category><![CDATA[cipher]]></category>
		<category><![CDATA[flat rate billing]]></category>
		<category><![CDATA[overruns]]></category>
		<category><![CDATA[pe web]]></category>
		<category><![CDATA[Plus]]></category>
		<category><![CDATA[product]]></category>
		<category><![CDATA[RATE]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32084</guid>
		<description><![CDATA[We are Broadvoice users some years ago, using Unlimited World PLUS product that seems reasonably acceptable. The problem is that we review Broadvoice config updates a lot of times to make our changes, but they are adding a lot of new countries to the flat rate (which is what alone is all that you want [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/broadvoice-unlimited-world-plus-dialplan-update-18may2012.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>50% of time SendDTMF failed</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/50-of-time-senddtmf-failed-2.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/50-of-time-senddtmf-failed-2.html#comments</comments>
		<pubDate>Fri, 18 May 2012 19:39:56 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[digit]]></category>
		<category><![CDATA[dtmf tones]]></category>
		<category><![CDATA[Ing]]></category>
		<category><![CDATA[tegui]]></category>
		<category><![CDATA[voip provider]]></category>
		<category><![CDATA[www api]]></category>
		<category><![CDATA[wwwww]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32082</guid>
		<description><![CDATA[I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected dynamically. Wanted to check with the community if this feature holds true on latest versions of Asterisk ? Regards, Mitul Limbani, Chief Architech &#038; Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai &#8211; 400 086. [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/50-of-time-senddtmf-failed-2.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>special digits * # on sip dial string</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/special-digits-on-sip-dial-string.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/special-digits-on-sip-dial-string.html#comments</comments>
		<pubDate>Fri, 18 May 2012 13:10:44 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[aditional]]></category>
		<category><![CDATA[call]]></category>
		<category><![CDATA[deliverd]]></category>
		<category><![CDATA[dial string]]></category>
		<category><![CDATA[information]]></category>
		<category><![CDATA[ngn]]></category>
		<category><![CDATA[ss7 network]]></category>
		<category><![CDATA[whas]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32067</guid>
		<description><![CDATA[hi guys. sorry if this is a silly question. My recharge application uses * digits if the subscriber wants to send some aditional information to speed up a process, dialing something like *777*123*5000 On my old ss7 network works great, but on my new ngn/sip i think it&#8217;s not possible because somewhere the call is [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/special-digits-on-sip-dial-string.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Best practices to route calls according holidays</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/best-practices-to-route-calls-according-holidays.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/best-practices-to-route-calls-according-holidays.html#comments</comments>
		<pubDate>Fri, 18 May 2012 12:57:34 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[button]]></category>
		<category><![CDATA[calendar resource]]></category>
		<category><![CDATA[private calendar]]></category>
		<category><![CDATA[private resource]]></category>
		<category><![CDATA[public calendar]]></category>
		<category><![CDATA[route]]></category>
		<category><![CDATA[toggle]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32065</guid>
		<description><![CDATA[Hi, At the moment, I&#8217;m mostly using a &#8220;Day/Night toggle&#8221; button to let users deal with week-ends, holidays and opening hours. As Asterisk 1.8 introduces Calendar capabilities, I&#8217;m wondering if better alternatives now exist. Is it possible, safe, reliable and easy to refer from Asterisk to a public calendar resource listing holidays, for a given [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/best-practices-to-route-calls-according-holidays.html/feed</wfw:commentRss>
		<slash:comments>2</slash:comments>
		</item>
		<item>
		<title>Transfer CDRs</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/transfer-cdrs.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/transfer-cdrs.html#comments</comments>
		<pubDate>Fri, 18 May 2012 12:32:48 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[call]]></category>
		<category><![CDATA[calldate]]></category>
		<category><![CDATA[constraint]]></category>
		<category><![CDATA[csv]]></category>
		<category><![CDATA[database table]]></category>
		<category><![CDATA[dst]]></category>
		<category><![CDATA[Field]]></category>
		<category><![CDATA[uniqueid]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32061</guid>
		<description><![CDATA[Hello, I&#8217;m using attended call transfer in asterisk 1.8.11.0 on a CentOS machine. Each CDR entry of calls that are transferred is repeated once. Every field including uniqueid, calldate, billsec, duration, src, dst, channel, dstchannel is exactly the same. Besides adding a constraint in the database table, isn&#8217;t there any way I can resolve this [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/transfer-cdrs.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>RTP stats explaination</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/rtp-stats-explaination.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/rtp-stats-explaination.html#comments</comments>
		<pubDate>Fri, 18 May 2012 11:00:35 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[clock drift]]></category>
		<category><![CDATA[drop packets]]></category>
		<category><![CDATA[max delta]]></category>
		<category><![CDATA[rtp packets]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32053</guid>
		<description><![CDATA[Hi all, This question is not related to asterisk, but related to voip quality in general. But i thought there are lot of experienced guys out here who can help me with this. And our telephony platform is also asterisk . May be i can extract some bias over this We are getting very poor [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/rtp-stats-explaination.html/feed</wfw:commentRss>
		<slash:comments>3</slash:comments>
		</item>
		<item>
		<title>Asterisk 1.8 canreinvite</title>
		<link>http://asteriskfaqs.org/2012/05/18/asterisk-users/asterisk-1-8-canreinvite.html</link>
		<comments>http://asteriskfaqs.org/2012/05/18/asterisk-users/asterisk-1-8-canreinvite.html#comments</comments>
		<pubDate>Fri, 18 May 2012 08:56:11 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[canreinvite]]></category>
		<category><![CDATA[directmedia]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32051</guid>
		<description><![CDATA[Hello, is canreinvite still supported in Asterisk 1.8 ?? I read about directmedia being available in asterisk 1.8, but is it the same ?? What happens when I use canreinvite in Asterisk 1.8 ?]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/18/asterisk-users/asterisk-1-8-canreinvite.html/feed</wfw:commentRss>
		<slash:comments>2</slash:comments>
		</item>
		<item>
		<title>Fax Problem on direct FXO port</title>
		<link>http://asteriskfaqs.org/2012/05/17/asterisk-users/fax-problem-on-direct-fxo-port.html</link>
		<comments>http://asteriskfaqs.org/2012/05/17/asterisk-users/fax-problem-on-direct-fxo-port.html#comments</comments>
		<pubDate>Thu, 17 May 2012 16:06:30 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[echo canceller]]></category>
		<category><![CDATA[mailing]]></category>
		<category><![CDATA[open source community]]></category>
		<category><![CDATA[port]]></category>
		<category><![CDATA[sending faxes]]></category>
		<category><![CDATA[soft switch]]></category>
		<category><![CDATA[spandsp]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32044</guid>
		<description><![CDATA[Sebastian, Seeing as this an issue related to faxing using the SpanDSP library; if you do not get an answer leading to a solution here, then you may try asking on the SpanDSP mailing list http://lists.soft-switch.org/mailman/listinfo It&#8217;s likely that the Asterisk users, specifically using SpanDSP, may be on that list. Thanks, Rusty Newton Open Source [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/17/asterisk-users/fax-problem-on-direct-fxo-port.html/feed</wfw:commentRss>
		<slash:comments>4</slash:comments>
		</item>
		<item>
		<title>realtime configuration for /etc/dahdi/system.conf</title>
		<link>http://asteriskfaqs.org/2012/05/17/asterisk-users/realtime-configuration-for-etcdahdisystem-conf.html</link>
		<comments>http://asteriskfaqs.org/2012/05/17/asterisk-users/realtime-configuration-for-etcdahdisystem-conf.html#comments</comments>
		<pubDate>Thu, 17 May 2012 12:38:07 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32028</guid>
		<description><![CDATA[Hi, can we load the settings of /etc/dahdi/system.conf from database table in real time. thanks,Kamlesh]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/17/asterisk-users/realtime-configuration-for-etcdahdisystem-conf.html/feed</wfw:commentRss>
		<slash:comments>1</slash:comments>
		</item>
		<item>
		<title>groups and categories</title>
		<link>http://asteriskfaqs.org/2012/05/17/asterisk-users/groups-and-categories.html</link>
		<comments>http://asteriskfaqs.org/2012/05/17/asterisk-users/groups-and-categories.html#comments</comments>
		<pubDate>Thu, 17 May 2012 10:02:15 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[brain farts]]></category>
		<category><![CDATA[call]]></category>
		<category><![CDATA[conference group]]></category>
		<category><![CDATA[COUNT]]></category>
		<category><![CDATA[GROUP]]></category>
		<category><![CDATA[group count]]></category>
		<category><![CDATA[inbound calls]]></category>
		<category><![CDATA[Potenial]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32024</guid>
		<description><![CDATA[I know that I should know this. But I&#8217;m having serious brain farts at the moment. I want to have a call be counted in a number of ways outbound inbound potential so, for example, a call comes into my dialplan, I want to add it to TotalCalls (all calls inbound + outbound) InboundCalls (all [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/17/asterisk-users/groups-and-categories.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Asterisk 10.1.3 on SUSE 10 SP2 &#8211; Resolved</title>
		<link>http://asteriskfaqs.org/2012/05/16/asterisk-users/asterisk-10-1-3-on-suse-10-sp2-resolved.html</link>
		<comments>http://asteriskfaqs.org/2012/05/16/asterisk-users/asterisk-10-1-3-on-suse-10-sp2-resolved.html#comments</comments>
		<pubDate>Wed, 16 May 2012 19:14:02 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[ASTCFLAGS]]></category>
		<category><![CDATA[asterisk]]></category>
		<category><![CDATA[fact]]></category>
		<category><![CDATA[fpic]]></category>
		<category><![CDATA[makefile]]></category>
		<category><![CDATA[Resolved]]></category>
		<category><![CDATA[suse 10]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32015</guid>
		<description><![CDATA[Hi List, The resolution for the problem was to modify utils/Makefile with this line _ASTCFLAGS+=-DSTANDALONE -fPIC Instead of _ASTCFLAGS+=-DSTANDALONE Still doesn&#8217;t resolve the fact that sqlite3 version 3.2.8 doesn&#8217;t like the create table if not exists astdb, but I can live with that. Thanks Danny Nicholas]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/16/asterisk-users/asterisk-10-1-3-on-suse-10-sp2-resolved.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>(no subject)</title>
		<link>http://asteriskfaqs.org/2012/05/16/asterisk-users/no-subject-9.html</link>
		<comments>http://asteriskfaqs.org/2012/05/16/asterisk-users/no-subject-9.html#comments</comments>
		<pubDate>Wed, 16 May 2012 17:07:06 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[Business]]></category>
		<category><![CDATA[Capet]]></category>
		<category><![CDATA[kevon]]></category>
		<category><![CDATA[letiste praha cz]]></category>
		<category><![CDATA[own your own business]]></category>
		<category><![CDATA[subject]]></category>
		<category><![CDATA[wingate]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32010</guid>
		<description><![CDATA[Generate $500  $2500 a month &#8211; Own Your Own Business http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329 ____________ Well, this is it, Capet. kevon wingate Wed, 16 May 2012 18:07:05]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/16/asterisk-users/no-subject-9.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>OT &#8211; Incoming fax cuts ADSL line</title>
		<link>http://asteriskfaqs.org/2012/05/16/asterisk-users/ot-incoming-fax-cuts-adsl-line.html</link>
		<comments>http://asteriskfaqs.org/2012/05/16/asterisk-users/ot-incoming-fax-cuts-adsl-line.html#comments</comments>
		<pubDate>Wed, 16 May 2012 16:07:31 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[adsl]]></category>
		<category><![CDATA[fax]]></category>
		<category><![CDATA[INCOMING]]></category>
		<category><![CDATA[incoming fax]]></category>
		<category><![CDATA[line]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=32005</guid>
		<description><![CDATA[]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/16/asterisk-users/ot-incoming-fax-cuts-adsl-line.html/feed</wfw:commentRss>
		<slash:comments>2</slash:comments>
		</item>
		<item>
		<title>50% of time SendDTMF failed</title>
		<link>http://asteriskfaqs.org/2012/05/16/asterisk-users/50-of-time-senddtmf-failed.html</link>
		<comments>http://asteriskfaqs.org/2012/05/16/asterisk-users/50-of-time-senddtmf-failed.html#comments</comments>
		<pubDate>Wed, 16 May 2012 15:07:05 +0000</pubDate>
		<dc:creator>Mailing-list Collector</dc:creator>
				<category><![CDATA[Asterisk-Users]]></category>
		<category><![CDATA[dtmf tones]]></category>
		<category><![CDATA[rfc2833]]></category>
		<category><![CDATA[uk thanks]]></category>
		<category><![CDATA[voip provider]]></category>

		<guid isPermaLink="false">http://asteriskfaqs.org/?p=31991</guid>
		<description><![CDATA[I am having a problem with SendDTMF() &#8211; 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(wwwww3wwwww2wwwwww1wwwww4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 work and failed to do number 2. Sometime all went [...]]]></description>
		<wfw:commentRss>http://asteriskfaqs.org/2012/05/16/asterisk-users/50-of-time-senddtmf-failed.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
	</channel>
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