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		<title>Asterisk Support</title>
		<link>http://voipforums.forumandco.com/asterisk-support-f11/-t1.htm</link>
		<description>Get help with installing, upgrading and running Asterisk.</description>
		<lastBuildDate>Tue, 21 Jul 2009 20:40:23 GMT</lastBuildDate>
		<ttl>10</ttl>
		<image>
			<title>Asterisk Support</title>
			<url>http://illiweb.com/fa/m/logo5.gif</url>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/-t1.htm</link>
		</image>
		<item>
			<title>VoipSwitch + Custom CDRs + MobileDialer in just $4000 by S4VOIP.com</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/voipswitch-custom-cdrs-mobiledialer-in-just-4000-by-s4voipcom-t450.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Dear Friends:



Want to setup VoIP company, a business under your own brand name? We have complete solution to launche VoIP (Voice Over Internet Protocol) company. All support comes included.

Features: PC2Phone, Device2Phone, Calling Card, Callback,sms callback solution, Ani Callback solution, DID callback solution, Cli Callback, Pin Callback, Wholesale Termination, Online Billing, Unlimited Resellers Creating, online shop, invoice generator, paypal integrated online shop, pin recharge modules,  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Tue, 21 Jul 2009 20:40:23 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/voipswitch-custom-cdrs-mobiledialer-in-just-4000-by-s4voipcom-t450.htm#453</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/voipswitch-custom-cdrs-mobiledialer-in-just-4000-by-s4voipcom-t450.htm</guid>
		</item>
		<item>
			<title>Looking for a simple way to launch telecommunication business?</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/looking-for-a-simple-way-to-launch-telecommunication-business-t437.htm</link>
			<dc:creator>thehosted</dc:creator>
			<description>Looking for a simple way to launch telecommunication businessLooking for a simple way to launch telecommunication business? ?</description>
			<category>Asterisk Support</category>
			<pubDate>Thu, 09 Jul 2009 21:16:55 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/looking-for-a-simple-way-to-launch-telecommunication-business-t437.htm#440</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/looking-for-a-simple-way-to-launch-telecommunication-business-t437.htm</guid>
		</item>
		<item>
			<title>The Complete Platform and Solution For Your Voip Business [sale/rent]</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/the-complete-platform-and-solution-for-your-voip-business-sale-rent-t436.htm</link>
			<dc:creator>thehosted</dc:creator>
			<description>Hello Clients,



Are you interested to start your own Voip business?  You are in the right place. We will provide you one solution that covers the whole range of your Voip Business requirements. We delivers SIP-based communications and services, 



Thehostedvoip offering best rent/sale solution with quality server for your Voip Business.



Features:



VoIP Termination

Wholesale VoIP

PC2Phone Platform

Device2Phone

Calling Cards Platform

Softphones VPN/Tunnel ( Customized  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Thu, 09 Jul 2009 21:15:47 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/the-complete-platform-and-solution-for-your-voip-business-sale-rent-t436.htm#439</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/the-complete-platform-and-solution-for-your-voip-business-sale-rent-t436.htm</guid>
		</item>
		<item>
			<title>!!! Start your Own Callshop, And wholesale business !!</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/start-your-own-callshop-and-wholesale-business-t418.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Dear Friends:



Want to setup VoIP company, a business under your own brand name? We have complete solution to launche VoIP (Voice Over Internet Protocol) company. All support comes included.

Features: PC2Phone, Device2Phone, Calling Card, Callback,sms callback solution, Ani Callback solution, DID callback solution, Cli Callback, Pin Callback, Wholesale Termination, Online Billing, Unlimited Resellers Creating, online shop, invoice generator, paypal integrated online shop, pin recharge modules,  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Thu, 02 Jul 2009 23:28:48 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/start-your-own-callshop-and-wholesale-business-t418.htm#421</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/start-your-own-callshop-and-wholesale-business-t418.htm</guid>
		</item>
		<item>
			<title>Extension based Outbound calling</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/extension-based-outbound-calling-t25.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Hi, 



I was not able to get this logic working. Below is my requirement. 



I have 2 PRI Links on my asterisk box. 



PRI#1 for US Incoming &amp; Outgoing 

PRI#2 for UK Incoming &amp; Outgoing 



I have 2 asterisk servers one at US and other at India connected over IAX2 trunking. (Private Leased lines - WAN). 



All extensions are getting registered at India Asterisk server. How do i make sure UK Extensions outgoing calls go thru UK PRI? is there any extensions (did) number  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:56:44 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/extension-based-outbound-calling-t25.htm#25</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/extension-based-outbound-calling-t25.htm</guid>
		</item>
		<item>
			<title>Did anybody has implemented SLA on Asterisk 1.4</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/did-anybody-has-implemented-sla-on-asterisk-14-t24.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Then please give the sample conf files. Following through the documentation on Digium. I managed to configure line1 and line2 for my Wildcard TDM400 and the incoming calls also creates pseudo-meetme SLATrunk_line1. But the phone configuration is ambiguous. 



Where I have to specify the phone configuration, I have done hints for (station1_line1) on Elmeg phone function keys and I have also tried configuring one of the sip extension with the channel as SIP/station1. But none of them work and  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:49:36 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/did-anybody-has-implemented-sla-on-asterisk-14-t24.htm#24</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/did-anybody-has-implemented-sla-on-asterisk-14-t24.htm</guid>
		</item>
		<item>
			<title>Call Transfer in PSTN</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/call-transfer-in-pstn-t23.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description><![CDATA[Asterisk works as a PBX. 
<br />

<br />
So can we configure it as follows: 
<br />

<br />
The asterisk has input PSTN lines and output also as PSTN lines. So it receives from PSTN and then it calls on another PSTN line and then transfers. Is that possible ?
<br />

<br />
Thanks]]></description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:46:48 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/call-transfer-in-pstn-t23.htm#23</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/call-transfer-in-pstn-t23.htm</guid>
		</item>
		<item>
			<title>How to analyze / log SIP on a production</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/how-to-analyze-log-sip-on-a-production-t22.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description><![CDATA[I have a production unit and there is loads of traffic I need to sort through, to figure out WHY just a few extensions will disconnect abruptly in the middle of a live call. So I'm looking for ways to trace or log specific extensions (or peers, phones, or whatever I can) 
<br />

<br />
Is there a way to log debug, error, or otherwise messages for one sip extension to a file? 
<br />

<br />
-Erich]]></description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:40:49 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/how-to-analyze-log-sip-on-a-production-t22.htm#22</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/how-to-analyze-log-sip-on-a-production-t22.htm</guid>
		</item>
		<item>
			<title>SLA Implementation using SIP Trunk</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/sla-implementation-using-sip-trunk-t21.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Good day everyone: 



I am trying to get SLA implemented in my office for testing purposes and to learn more about it. We are using Trixbox, which is a flavor of Asterisk, with both Linksys SPA942 phones and Aastra 57ict phones. All of our incoming lines are SIP trunks hosted with Teliax with NO ZAP cards or FXO boards. 



Based on reading a ton of posts and PDF documents, this is what I have so far: 



*************SLA.CONF********** 

[general] 

attemptcallerid=no 

ringtimeout=30  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:39:16 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/sla-implementation-using-sip-trunk-t21.htm#21</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/sla-implementation-using-sip-trunk-t21.htm</guid>
		</item>
		<item>
			<title>registeration of sip</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/registeration-of-sip-t20.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Hi 

My call between to sip mode is working allright but when ever iam opening my X-lite on CLI iam getting this msg 



Code: 

Registered SIP '2000' at 192.168.1.40 port 5062 expires 1800 

[Jul 11 16:15:23] NOTICE[3342]: chan_sip.c:12599 handle_response_peerpoke: Peer '2000' is now Reachable. (2ms / 2000ms) 





what does this message ,means Registered SIP '2000' at 192.168.1.40 port 5062 expires 1800 



I feel this msg shows that the number 2000 is registered but when iam giving  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:37:51 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/registeration-of-sip-t20.htm#20</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/registeration-of-sip-t20.htm</guid>
		</item>
		<item>
			<title>How to analyze / log SIP on a production</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/how-to-analyze-log-sip-on-a-production-t19.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description><![CDATA[I have a production unit and there is loads of traffic I need to sort through, to figure out WHY just a few extensions will disconnect abruptly in the middle of a live call. So I'm looking for ways to trace or log specific extensions (or peers, phones, or whatever I can) 
<br />

<br />
Is there a way to log debug, error, or otherwise messages for one sip extension to a file?]]></description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:35:38 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/how-to-analyze-log-sip-on-a-production-t19.htm#19</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/how-to-analyze-log-sip-on-a-production-t19.htm</guid>
		</item>
		<item>
			<title>phone issue,Pls support</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/phone-issuepls-support-t18.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>Alright, I am sort of a noob but I am excited to get Asterisk up and running for my home office. I seem to have ran into a couple of holes that I can't climb out of... 



I have an SPA941 phone with 2 line appearances. Even though I have configured the lines as separate extensions in the web interface, regardless of which one rings line one lights up. Is this normal? Also, is there a way to make this like the good old analog pbx phone so when can select a line the call goes out the corresponding  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:34:56 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/phone-issuepls-support-t18.htm#18</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/phone-issuepls-support-t18.htm</guid>
		</item>
		<item>
			<title>dial plan solution</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/dial-plan-solution-t17.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description>I have a installed asterisk from source and it is all working correctly, but i have a dialplan question. Our reception ext is 100 however the reception phone is actually ext162. The 100 ext just forwards to the reception phone. If someone rings 162 directly (and the receptionist is on the phone) it plays a message saying they are on the phone. If someone dials 100 it diverts 

to 162 but if theye are on the phone it doesn't play the message. I have pasted the part of my dialplan that tells 100  ...</description>
			<category>Asterisk Support</category>
			<pubDate>Sat, 12 Jul 2008 07:31:05 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/dial-plan-solution-t17.htm#17</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/dial-plan-solution-t17.htm</guid>
		</item>
		<item>
			<title>need asteriks professionals</title>
			<link>http://voipforums.forumandco.com/asterisk-support-f11/need-asteriks-professionals-t5.htm</link>
			<dc:creator>Anonymous</dc:creator>
			<description><![CDATA[Hi all,
<br />

<br />
we need asterisk professionals
<br />

<br />
regards
<br />
<a href="mailto:voipsoftphone@yahoo.co.in">voipsoftphone@yahoo.co.in</a>]]></description>
			<category>Asterisk Support</category>
			<pubDate>Fri, 11 Jul 2008 08:00:20 GMT</pubDate>
			<comments>http://voipforums.forumandco.com/asterisk-support-f11/need-asteriks-professionals-t5.htm#5</comments>
			<guid>http://voipforums.forumandco.com/asterisk-support-f11/need-asteriks-professionals-t5.htm</guid>
		</item>
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