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    <title>Grandstream</title>
    <link>http://forum.freespeech.ie/list/3</link>
    <description><![CDATA[Discussion of Grandstream VOIP hardware]]></description>
    <language>EN</language>
    <pubDate>Thu, 14 Jun 2007 14:37:46 +0100</pubDate>
    <lastBuildDate>Thu, 14 Jun 2007 14:37:46 +0100</lastBuildDate>
    <category>Grandstream</category>
    <generator>Phorum 5.1.11</generator>
    <ttl>600</ttl>
    <item>
      <title>Re: Grandstream GXP2000 Not Ringing</title>
      <link>http://forum.freespeech.ie/read/3/271/274#msg-274</link>
      <author>moderator</author>
      <description><![CDATA[I suggest trying the softphone. 

It may be that your firewall is blocking the SIP messages. If you phone is unreachable but the contact has been updated to the service then it will continue to try for a period before going to voicemail... if you hear ringing for an extended period (&gt;30 secs) then your phone must be generating the tone.]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/271/274#msg-274</guid>
      <pubDate>Thu, 14 Jun 2007 14:37:46 +0100</pubDate>
    </item>
    <item>
      <title>Re: Grandstream GXP2000 Not Ringing</title>
      <link>http://forum.freespeech.ie/read/3/271/273#msg-273</link>
      <author>damiens</author>
      <description><![CDATA[Hi Mod,

have not tried softphone.

I'm calling from a mobile to the unit.]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/271/273#msg-273</guid>
      <pubDate>Thu, 14 Jun 2007 12:48:10 +0100</pubDate>
    </item>
    <item>
      <title>Re: Grandstream GXP2000 Not Ringing</title>
      <link>http://forum.freespeech.ie/read/3/271/272#msg-272</link>
      <author>moderator</author>
      <description><![CDATA[1. Does the software phone work ? 
2. Where is the call coming from, is it another SIP phone or from the PSTN ?]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/271/272#msg-272</guid>
      <pubDate>Thu, 14 Jun 2007 12:24:01 +0100</pubDate>
    </item>
    <item>
      <title>Grandstream GXP2000 Not Ringing</title>
      <link>http://forum.freespeech.ie/read/3/271/271#msg-271</link>
      <author>damiens</author>
      <description><![CDATA[Hi,

I currently have a fortinet 60a firewall and Grandstream GXP2000.

I can call out fine, but if someone rings me, it does not ring on the GXP2000 but ringing tone comes from the dialling phone.

has anybody has this issue?]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/271/271#msg-271</guid>
      <pubDate>Wed, 13 Jun 2007 19:48:28 +0100</pubDate>
    </item>
    <item>
      <title>Re: Grandstream ATA Set up</title>
      <link>http://forum.freespeech.ie/read/3/148/149#msg-149</link>
      <author>moderator</author>
      <description><![CDATA[The ports you need to have open on Outbound Policy (towards the internet)
=========================================================================
A residential firewall is normally configured to allow any ports on Outbound.

SIP (signalling):5060 (UDP &amp; TCP)
STUN (NAT-TRAVERSAL if you use): 3478 (UDP)
RTP (media/voice) - this is dependant, normally in the range from 5000 - 1000 (UDP). If your talking to other SIP clients then it's peer-2-peer so it's dependant on the range they use. 

The ports you need to have open on Inbound Policy (from internet)
=========================================================================
These ports are the ones you configure on the phone. 

This default are normally 

SIP:5060 (UDP &amp; TCP)
RTP:5000-1000 (UDP)]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/148/149#msg-149</guid>
      <pubDate>Sat, 30 Sep 2006 11:20:10 +0100</pubDate>
    </item>
    <item>
      <title>Grandstream ATA Set up</title>
      <link>http://forum.freespeech.ie/read/3/148/148#msg-148</link>
      <author>Fudskie</author>
      <description><![CDATA[For you to use your Grandstream ATA you need to fill in these sections of the menu within the Adapter.

Sip Server = freespeech.ie
Sip User ID = your Freespeech number 07 etc etc
Authenticate ID = same as above (Sip user ID) 07**** etc
Authenticate Password = your account password
Leave all other fields as the default.

Scroll down the menus until you find:
User ID is phone number: No / Yes (select yes)
Use random port: No / Yes  (select Yes)
NAT Traversal: No (even if you are NAT leave this as no)

As i have yet to find out the exact ports this network uses i have set my Grandstream to have a fixed internal IP address. I have then set up a DMZ in the router to point to this internal address the Grandstream is on. I have yet to find all the ports that this network uses so i have had to set the DMZ. If any one knows the total port range Freespeech use i would be grateful if you could post them here.

Thx Fudskie......]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/148/148#msg-148</guid>
      <pubDate>Sat, 30 Sep 2006 11:09:08 +0100</pubDate>
    </item>
    <item>
      <title>Grandstream HT-488</title>
      <link>http://forum.freespeech.ie/read/3/62/62#msg-62</link>
      <author>david.marshall</author>
      <description><![CDATA[Hi anyone and everyone
I've bought a couple of Grandstream HT-488 to use with my Freespeech account. This features 1 FXS and 1 FXO port for call forwarding between VoIP and PSTN. As well as PSTN pass-through so I can make PSTN or VoIP calls from one handset.

I'm getting a couple of problems.  I'll just deal with the first here.

I get line echo when I make a PSTN call.  I suspect it's to do with one, or both of two settings in the FXS port.  These are:

1) FXS impedance, which I have set at CTR21 (270 Ohm + 750 Ohm || 150nF). Does anyone know what I should choose? 
2) Oh hook voltage 36V.  I guess this depends on Eircom.  Again any suggestions on what it should be.

And 3) I have Caller ID set at ETSI-FSK... should it be ETSI-DTMF does anyone know?

If I can get these settings sorted, I can see if addresses some of the other issues experienced by callers.

Hope someone out there knows

Regards
David Marshall]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/62/62#msg-62</guid>
      <pubDate>Thu, 18 May 2006 15:12:59 +0100</pubDate>
    </item>
    <item>
      <title>web link</title>
      <link>http://forum.freespeech.ie/read/3/23/23#msg-23</link>
      <author>moderator</author>
      <description><![CDATA[http://www.grandstream.com/]]></description>
      <category>Grandstream</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/3/23/23#msg-23</guid>
      <pubDate>Sun, 23 Apr 2006 22:30:43 +0100</pubDate>
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