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	<title><![CDATA[AudioCodes Forums]]></title>
	<link>http://audiocodes.websitetoolbox.com</link>
	<description><![CDATA[AudioCodes Forums]]></description>
	<ttl>60</ttl>
	<pubDate>Wed, 16 May 2012 14:28:23 GMT</pubDate>
	<item>
		<title><![CDATA[BRI problem]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5806153</link>
		<description><![CDATA[<p dir="ltr"><font size="3"><font face="Times New Roman"><span style='color: rgb(32, 20, 20); line-height: 115%; font-family: "Verdana","sans-serif"; font-size: 8.5pt;'>I have a BRI card in my Mediant 800 and Ihave 4 BRI's connecting to service provider. It should be 8 concurrent outboundcalls but we’ve only got 2.</span></font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font face="Times New Roman"><font size="3"><span style='color: rgb(32, 20, 20); line-height: 115%; font-family: "Verdana","sans-serif"; font-size: 8.5pt;'>Here is the status of the BRI ports. Whenthere is no call, the ports turn red. But when there is a call, the port turnsgreen. The device only selects port 1 and port 2 for its outbound calls. But forinbound calls, the service provider would select any of the BRI ports.</span><span style='line-height: 115%; font-family: "Calibri","sans-serif"; font-size: 11pt; mso-fareast-font-family: SimSun; mso-bidi-font-family: "Times New Roman"; mso-ansi-language: EN-AU; mso-fareast-language: ZH-CN; mso-bidi-language: AR-SA;'></span></font><font size="3"></font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3"><font face="Calibri">                             </font></font><font size="3"><font face="Calibri"> </font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3"><font face="Calibri">Below is the configuration for the trunk.</font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3"><font face="Calibri"> </font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3"><font face="Calibri"> </font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3"><font face="Calibri"> </font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Calibri">&nbsp;</font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3"><font face="Calibri"> </font></font></p><p style="margin: 0cm 0cm 10pt;" class="MsoNormal"><font size="3" face="Times New Roman"></font></p> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5806153</guid>
		<pubDate>Fri, 20 Apr 2012 03:27:49 GMT</pubDate>
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		<title><![CDATA[MP118 Call Problem]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5734940</link>
		<description><![CDATA[We have 2 MP118. Barnk office and head quarter.<br><br>The wproblem is I can not call 2 line at the same time.<br><br>Second line reings from branch office but the voice is one way audio.<br><br>Can not hear branch offices voice.<br><br>What is the problem?<br><br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5734940</guid>
		<pubDate>Fri, 02 Mar 2012 11:10:32 GMT</pubDate>
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	<item>
		<title><![CDATA[MP-118 cannot outcall]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5687000</link>
		<description><![CDATA[I cannot get the outcalls to work. When I dial out the call gets answered by the gateway but does not dial to the originating dialed number through one of the pstn lines.<br><br><br><br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5687000</guid>
		<pubDate>Tue, 31 Jan 2012 13:52:26 GMT</pubDate>
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		<title><![CDATA[Mediant 1000 - Busy out PRI]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5659044</link>
		<description><![CDATA[Does anyone have a method of busying out the voice channels of a PRIs that does not terminate active calls?&nbsp; I'm looking for a way to quiesce the PRIs so the Mediant can be taken out of service, rebooted, etc.<br><br><br><br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5659044</guid>
		<pubDate>Thu, 12 Jan 2012 14:32:21 GMT</pubDate>
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		<title><![CDATA[AudioCode MP114 Caller ID]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5594006</link>
		<description><![CDATA[hi,<br>I got an audiocode that I configure with my asterix server. every thing is ok but when I receive a call for external it did not display the caller id of the external caller but the number of the sip count that I introduce in the Audiocode. how can I fix it?? please help me<br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5594006</guid>
		<pubDate>Thu, 24 Nov 2011 15:23:05 GMT</pubDate>
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		<title><![CDATA[MP118 WebInterface stops at "Loading in Progress" after Firmware Upgrade]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5488627</link>
		<description><![CDATA[<P>Hello everybody,<BR><BR>I've run into a problem while upgrading the firmware on our MP118s. On the first device, everything went smoothly when upgrading to&nbsp;Firmware 6.20A.<BR><BR>On the backup device I did upgrade to Firmware 6.20A via the update wizard. I uploaded the Firmware&nbsp;cmp file and kept the rest&nbsp;(configuration file etc.) in the original state - just like with the previous device.<BR><BR>Once the firmware was burnt into the flash the device rebooted. When I tried to connect to the web interface, i got prompted for username and password and now the device keeps displaying the "Loading In Progress..." message indefinitely.<BR><BR>I tried a factory reset, which changed the IP address, username and password&nbsp;back to default - the problem with the web interface remains however.<BR><BR>I'd appreciate any suggestions,</P>Best regards,<BR>David<BR><br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5488627</guid>
		<pubDate>Tue, 13 Sep 2011 07:01:50 GMT</pubDate>
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		<title><![CDATA[Tandem Switching Mediant 2000]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5410294</link>
		<description><![CDATA[Hi Everyone,<div><br></div><div>Has anyone configured how to do tandem switching on Mediant?</div><div><br></div><div><br></div> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5410294</guid>
		<pubDate>Mon, 25 Jul 2011 08:02:14 GMT</pubDate>
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		<title><![CDATA[MP-114 wrong dial out]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5395548</link>
		<description><![CDATA[<P><SPAN style="FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US" lang=EN-US>Hi, I'm having a wired problem. For lab testing and demo purposing I set up a MP-114 FXO GW with Lync 2010. Everything went well until we tried to dial out to PTSN.</SPAN></P><SPAN style="FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US" lang=EN-US>Calls cannot be completed except those beginning with “5” (five). Actually I realized that the GW is changing the first digit somehow, so I get a wrong number or a busy signal. I'm not making any digits manipulation. In the LOG I see the special digit "e" automatically added to the destination phone number:</SPAN><BR><SPAN style="FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US" lang=EN-US>NOTICE&nbsp; : (&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; lgr_flow)(483&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; )&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; #0<img src="/images/boards/smilies/biggrin.gif" border="0" align="absmiddle">ialPhoneNumber=e42957024 <BR>NOTICE&nbsp; : (&nbsp;&nbsp; lgr_psbrdif)(482&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; )&nbsp; #0 Sending DTMFs (e42957024, Dir=2, Len=9, InterTime=100, DigitLen=100)<BR></SPAN><BR><SPAN style="FONT-FAMILY: 'Verdana','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US" lang=EN-US>Behavior changes if I disable the “</SPAN><SPAN style="FONT-FAMILY: 'Arial-BoldMT','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US; mso-bidi-font-family: Arial-BoldMT; mso-bidi-font-weight: bold" lang=EN-US>EnableMicrosofExt” option. In that case, no call completes. </SPAN><BR><P><SPAN style="FONT-FAMILY: 'Arial-BoldMT','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US; mso-bidi-font-family: Arial-BoldMT; mso-bidi-font-weight: bold" lang=EN-US>I tried several parameters changes, always getting same result. </SPAN></P><P><SPAN style="FONT-FAMILY: 'Arial-BoldMT','sans-serif'; FONT-SIZE: 10pt; mso-ansi-language: EN-US; mso-bidi-font-family: Arial-BoldMT; mso-bidi-font-weight: bold" lang=EN-US>Your help will be really appreciated. Thanks in advance. </SPAN></P><P style="MARGIN: 0cm 0cm 0pt" class=MsoNormal><SPAN style="COLOR: #1f497d; mso-ansi-language: EN-US; mso-ascii-font-family: Calibri; mso-hansi-font-family: Calibri; mso-bidi-font-family: Calibri" lang=EN-US><FONT size=3 face=Calibri>&nbsp;</FONT></SPAN></P><P>&nbsp;</P> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5395548</guid>
		<pubDate>Fri, 15 Jul 2011 19:57:14 GMT</pubDate>
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		<title><![CDATA[Monitoring ATA-MP114 Registered status through SNMP]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5354670</link>
		<description><![CDATA[Greetings,<br><br>I am using ATA MP-114 and I configured them to use SIP and to register per endpoint.<br><br>I am trying to monitor the endpoints' registration status.<br>It is very easy on the HTTP interface, but I can't find which OID to use to get this information through SNMP.<br><br>Does anybody know which OID to use ?<br><br>Thank you for your support.<br><br>Regards,<br>DOUILLE Tristan<br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5354670</guid>
		<pubDate>Wed, 15 Jun 2011 15:28:46 GMT</pubDate>
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		<title><![CDATA[mediant 1000 no incoming call]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5268766</link>
		<description><![CDATA[Hi,<div><br></div><div>I just want to ask if somebody can tell me how to configure incoming calls?</div><div>I have a mediant 1000 with 1 Trunk card and 1x4 FXS card. Outgoing call has no problem, but I can't figure out how to obtain an incoming call.</div><div><br></div><div>I tried a lot of manipulations but non of it worked.</div><div>I always got "Abnormal Disconnect cause:21#GWAPP_CALL_REJECTED Trunk:0 Conn:255".</div><div><br></div><div>Attached is the topology, Configuration, and some test logs.</div><div>The telephone number of E1 is 8587500 ISDN EURO.</div><div><br></div><div>Hope somebody can help me.</div><div><br></div><div>Thanks.</div> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5268766</guid>
		<pubDate>Sun, 22 May 2011 07:40:29 GMT</pubDate>
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		<title><![CDATA[MP-202 B F2XS: SIP register isn't automatic]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5260000</link>
		<description><![CDATA[Hello,<br>I have a problem with the SIP registration on the ATA MP-202 B 2FXS<br>For example, after uploading my configuration file, the line of my ATA will have the status "Not registred" on "Sip registration".<br>I must press OK on "Signaling protocol" tab on "VoIP" field to register my line.<br>I wish it would be done by itself.<br>How I can set up this?<br>For your information I'm editing directly the main conf file.<br><br>Well it's a point but I have another problem :<br>In my conf file i chosed to unencrypt the password.<br><br>So i use this command : (auth_password({“toto”})) on "line parameters" on "VoIP" field.<br>I found this command in the tutorial for the last software version (the one i'm using) but it doesn't work,<br>my CA doesn't recognize this password<br><br>I checked the conf on MP202B interface and it shows me this : (auth_password(toto))<br>So I supose it understood that the password isn't encrypt but it not doing the hash by next.<br><br>Just to clarify, the problem doesn't come from my CA because with the crypted password it's work.<br><br>But I'm doing an Excel in which the customer can type the password he wants and that transcribe it in the line (auth_password({"the_password_typed"}))<br><br>Even i have no way to crypt the password typed, can you tell me why the syntax {"password"} doesn't work for un-encrypt it ?<br><br>I would be very grateful if someone could help me.<br><br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5260000</guid>
		<pubDate>Tue, 17 May 2011 15:37:05 GMT</pubDate>
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		<title><![CDATA[Audiocodes mp-118 will not save a deafult gateway]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5216302</link>
		<description><![CDATA[<P>By using the inbuilt voice control I can set the IP and the subnet mask, however, when i try to set the default gateway, after pressing 1 on the phone for "save" it ALWAYS resets to 0.0.0.0. Any ideas, or do I have a faulty device?</P> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5216302</guid>
		<pubDate>Wed, 04 May 2011 11:54:27 GMT</pubDate>
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		<title><![CDATA[MP124 and TLS]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5200905</link>
		<description><![CDATA[Anyone have success using TLS on a MP-124?&nbsp; If so, can you post the configuration settings you used?&nbsp; Thanks.<br> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5200905</guid>
		<pubDate>Sat, 23 Apr 2011 14:37:07 GMT</pubDate>
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		<title><![CDATA[MP-202B VLANs]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5152131</link>
		<description><![CDATA[Hello All!<BR><BR>Does&nbsp;MP-202B support VLANs?<BR>When I configure this device to use separate VLAN for data and voice only one of there is working (either a voice, or the data).<BR><BR>When<BR>Data VLAN configured NAPT mode, metric = 1, default gateway checked and<BR>Voice&nbsp;VLAN configured&nbsp;ROUTE mode, metric = 0, default gateway checked only VoIP is working <IMG border=0 align=absMiddle src="http://audiocodes.websitetoolbox.com/images/boards/smilies/confused.gif"><BR>If for data VLAN set metric = 0 and for Voice VLAN set metric = 1 then data is working <IMG border=0 align=absMiddle src="http://audiocodes.websitetoolbox.com/images/boards/smilies/mad.gif"><BR><BR><B>What options to&nbsp;set that it worked together?</B><BR><BR>Thank you for information!<BR><BR> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5152131</guid>
		<pubDate>Wed, 23 Mar 2011 15:46:56 GMT</pubDate>
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		<title><![CDATA[Set SIP Contact on a MP-112 FXS]]></title>
		<link>http://audiocodes.websitetoolbox.com/post?id=5149302</link>
		<description><![CDATA[Hi<div><br></div><div>I have a&nbsp;MP-112 FXS where I need the to add the port number in the SIP reg Contact. Something like this:</div><div>Contact: &lt;sip:xxxx@aaa.bbb.ccc.ddd:5060;transport=udp&gt;</div><div><br></div><div>How du I do that?</div><div><br></div><div>Thanks</div><div><br></div><div>Lasse</div> <p>Forum: <a href="http://audiocodes.websitetoolbox.com/?forum=128277">Microsoft Forum</a>
]]></description>
		<guid isPermaLink="false">http://audiocodes.websitetoolbox.com/post?id=5149302</guid>
		<pubDate>Mon, 21 Mar 2011 21:29:26 GMT</pubDate>
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