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	<title>Roachys Weblog &#187; Asterisk</title>
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		<title>Roachys Weblog &#187; Asterisk</title>
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		<title>DTMF Tones not audible using IAX2 Trunk</title>
		<link>http://blog.roachy.net/2009/06/25/dtmf-tones-not-audible-using-iax2-trunk/</link>
		<comments>http://blog.roachy.net/2009/06/25/dtmf-tones-not-audible-using-iax2-trunk/#comments</comments>
		<pubDate>Thu, 25 Jun 2009 07:33:16 +0000</pubDate>
		<dc:creator>Paul Morgan-Roach</dc:creator>
				<category><![CDATA[Linux]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[DTMF]]></category>
		<category><![CDATA[IAX]]></category>
		<category><![CDATA[IAX2]]></category>
		<category><![CDATA[UDP]]></category>
		<category><![CDATA[VoIP]]></category>

		<guid isPermaLink="false">http://technicalmumblings.wordpress.com/?p=154</guid>
		<description><![CDATA[IAX2 DTMF tones being blocked, Asterisk.<img alt="" border="0" src="http://stats.wordpress.com/b.gif?host=blog.roachy.net&amp;blog=2880390&amp;post=154&amp;subd=technicalmumblings&amp;ref=&amp;feed=1" width="1" height="1" />]]></description>
			<content:encoded><![CDATA[<p>We&#8217;ve had a recent issue with IAX2 trunks whereby any DTMF tones played locally are not audible at the remote side of the connection&#8230;</p>
<p>Interestingly tones were audible on inbound and internal calls, however, this means that IVR&#8217;s are completely non-navigable.</p>
<p>The problem turned out to be due to the fact that it appears DTMF traffic was being sent out over a separate UDP port to the rest of the IAX traffic&#8230;.calls sounded fine, but DTMF traffic was being blocked due to it running on port 4571.  We&#8217;ve opened the range 4569-4571 and now all is working fine&#8230;.</p>
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			<media:title type="html">roachy1979</media:title>
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		<title>Trixbox &#8211; &#8220;All Circuits are Busy Now&#8221; when Dialling Off-Hook</title>
		<link>http://blog.roachy.net/2009/05/11/trixbox-all-circuits-are-busy-now-when-dialling-off-hook/</link>
		<comments>http://blog.roachy.net/2009/05/11/trixbox-all-circuits-are-busy-now-when-dialling-off-hook/#comments</comments>
		<pubDate>Mon, 11 May 2009 12:08:45 +0000</pubDate>
		<dc:creator>Paul Morgan-Roach</dc:creator>
				<category><![CDATA[Uncategorized]]></category>
		<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Trixbox]]></category>

		<guid isPermaLink="false">http://technicalmumblings.wordpress.com/?p=147</guid>
		<description><![CDATA[How to solve Trixbox - "All Circuits are Busy Now" when Dialling Off-Hook<img alt="" border="0" src="http://stats.wordpress.com/b.gif?host=blog.roachy.net&amp;blog=2880390&amp;post=147&amp;subd=technicalmumblings&amp;ref=&amp;feed=1" width="1" height="1" />]]></description>
			<content:encoded><![CDATA[<p>We have some Polycom IP 330 SIP handsets connected to a Trixbox.  Unfortunately, when using the default SIP.cfg that is downloaded via TFTP to the handsets, if the phone was off the hook (ie. a dial tone was already audible) the handsets would appear to time-out and give the message &#8220;All Circuits are Busy Now&#8221;. On looking at the call in the CLI (asterisk -rvvvv) it was apparent that only the first 9 digits were being dialled.</p>
<p>The solution was to edit the the /tftpboot/sip.cfg and look for the line that says:</p>
<p style="padding-left:30px;">&lt;digitmap dialplan.digitmap=&#8221;[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT&#8221; dialplan.digitmap.timeOut=&#8221;3|3|3|3|3|3&#8243;/&gt;</p>
<p>and replace with the correct number of digits for calls in your country&#8230;</p>
<p style="padding-left:30px;">&lt;digitmap dialplan.digitmap=&#8221;[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxxxx|[2-9]xxxxxxxxxxx|[2-9]xxxT&#8221; dialplan.digitmap.timeOut=&#8221;3|3|3|3|3|3&#8243;/&gt;</p>
<p>You then need to restart the handsets in order for this to work.</p>
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		<slash:comments>7</slash:comments>
	
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			<media:title type="html">roachy1979</media:title>
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