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  <channel>
    <title>Asterisk</title>
    <link>http://forum.freespeech.ie/list/14</link>
    <description><![CDATA[Discussion of Asterisk VOIP software]]></description>
    <language>EN</language>
    <pubDate>Sat, 25 Oct 2008 11:19:23 +0100</pubDate>
    <lastBuildDate>Sat, 25 Oct 2008 11:19:23 +0100</lastBuildDate>
    <category>Asterisk</category>
    <generator>Phorum 5.1.11</generator>
    <ttl>600</ttl>
    <item>
      <title>Re: Trixbox and DTMF Tones</title>
      <link>http://forum.freespeech.ie/read/14/294/339#msg-339</link>
      <author>miken1957</author>
      <description><![CDATA[I have just moved over to Freespeech.co.uk and have exactly the same problem with DTMF tones not being recognised by my IVR's, although other sip trunks like sipgate and voipfone work fine. Would you mind listing your Trixbox Freespeech trunk details so I can do the same, assuming you did crack the problem. thanks in advance]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/294/339#msg-339</guid>
      <pubDate>Sat, 25 Oct 2008 11:19:23 +0100</pubDate>
    </item>
    <item>
      <title>Re: Trixbox 2.2 / Asterisk 1.2.8</title>
      <link>http://forum.freespeech.ie/read/14/280/316#msg-316</link>
      <author>macchonmhaighe</author>
      <description><![CDATA[The same thing is happening to me. Did you manage to get this resolved?

I had inbound and outbound working (outbound had a few issues calling 076 numbers but was working for everything else)
After reinstalling I cant get inbound anymore.]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/280/316#msg-316</guid>
      <pubDate>Tue, 30 Oct 2007 10:57:09 +0000</pubDate>
    </item>
    <item>
      <title>Re: Trixbox and DTMF Tones</title>
      <link>http://forum.freespeech.ie/read/14/294/295#msg-295</link>
      <author>moderator</author>
      <description><![CDATA[What methods of DTMF are you supporting and what codec's are you using 

post your SIP config]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/294/295#msg-295</guid>
      <pubDate>Wed, 18 Jul 2007 14:50:20 +0100</pubDate>
    </item>
    <item>
      <title>Trixbox and DTMF Tones</title>
      <link>http://forum.freespeech.ie/read/14/294/294#msg-294</link>
      <author>Alan Roche</author>
      <description><![CDATA[Hi everyone

I am having a problem with DTMF tones on an Asterisk / Trixbox , basically its not working with Freespeech.ie. It works perfectly internally but not external when i call in. If anyone has any information or solution PLEASE give shout on this.

Thanks everyone 

Al.]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/294/294#msg-294</guid>
      <pubDate>Wed, 18 Jul 2007 12:08:51 +0100</pubDate>
    </item>
    <item>
      <title>Trixbox 2.2 / Asterisk 1.2.8</title>
      <link>http://forum.freespeech.ie/read/14/280/280#msg-280</link>
      <author>david.marshall</author>
      <description><![CDATA[Hi
I'm looking for some help configuring an asterisk 1.2.8/SIP connection with Freespeech.

Does anyone have an Asterisk set-up I can try other than the one here on the forum...

Some background if you are interested:

I have had a successful Asterisk@home/Trixbox 1.x connection to Freespeech for 13 months.  No issues.

Following a hard drive failure I upgraded the device and installed Trixbox 2.2.  Actually there was so much to learn, and future recovery to discover that I actually build several versions of Trixbox 2.2, and asterisk 1.2.18, 1.2.19 and 1.4.1 before doing a clean install of Trixbox 2.2 with asterisk 1.2.18.

With all of these installs I have been able to succesfully create trunk SIP connections to a variety of iTSPs ..except Freespeech. But I want to use Freespeech because we generally find that the call quality is better... our qualify results are certaily much better than others.

And I do not know why we are failing.  I've followed the setup format here on this and the .co.uk forums, I've used sip debug peer freespeech to try to see what is happening.. I've used FreePBX to config and I've edited the conf files directly.

The asterisk CLI shows clean SIP headers, but the asterisk log shows them deformed.

We could dial in, but after all the attemted debugging we can't even do that.

Any ideas would be appreciated

Regards
David]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/280/280#msg-280</guid>
      <pubDate>Thu, 28 Jun 2007 12:48:53 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/270#msg-270</link>
      <author>moderator</author>
      <description><![CDATA[It's likely you do not have your inbound context/dialplan correctly defined. 
in sip.conf you have define the context where the calls comes in 

context=(your inbound context)

You need to make sure you have a dialplan set up correctly in this context for receiving the call.


hint: Enter the asterisk command line (asterisk -rvvvvv
) and you should be able to debug the problem when making a call. If you want to see the call coming in then run &gt;sip debug to trace the messages.]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/270#msg-270</guid>
      <pubDate>Mon, 11 Jun 2007 09:55:24 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/269#msg-269</link>
      <author>Alan Roche</author>
      <description><![CDATA[Hi Everyone

just wondering if some one could give me a hand with setting up Asterisk with Freespeech. i have got it dialing out through freespeech but i am unable to receive incoming calls from it, every time i ring from my mobile i get freespeech's voice mail. i have added a dialing rule and also an incoming dialing rule from freespeech.ie but still nothing. 

i am only using the free account for now till i am satisfied that Asterisk will work correctly. I followed the instructions from this forum and managed to get it to dial out but as i said nothing in :) 

Thanks in advance for the help on this

Al.

Ps on the freespeech.ie website it does say that i am online]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/269#msg-269</guid>
      <pubDate>Sun, 10 Jun 2007 11:10:51 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/184#msg-184</link>
      <author>pratheesh</author>
      <description><![CDATA[Hi,
I am using Trixbox version 2.
I added the above settings via the 'Trunks' Setup menu.
'Asterisk Info' page shows that the sip peer is registered.
Should i be able to dial out using x-lite if i topup my account or is there anything else that i need to setup.

***************
Sip Peers

Host                            Username       Refresh State               
freespeech.ie:5060              076XXXXXXX         105 Registered          
Verbosity is at least 1
***************

Thanks guys]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/184#msg-184</guid>
      <pubDate>Tue, 20 Feb 2007 16:45:59 +0000</pubDate>
    </item>
    <item>
      <title>Re: Trixbox v2</title>
      <link>http://forum.freespeech.ie/read/14/182/183#msg-183</link>
      <author>moderator</author>
      <description><![CDATA[take a look at this thread - http://forum.freespeech.ie/read/14/51]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/182/183#msg-183</guid>
      <pubDate>Tue, 20 Feb 2007 15:19:29 +0000</pubDate>
    </item>
    <item>
      <title>Trixbox v2</title>
      <link>http://forum.freespeech.ie/read/14/182/182#msg-182</link>
      <author>pratheesh</author>
      <description><![CDATA[Hi,
 I have installed the trixbox version 2 using the prebuilt image.
Could anyone tell me how i can configure it to use my freespeech account for dialing out and receiving calls? 

I have setup x-lite softphones on two pc's and have created internal extensions for them on trixbox. They work properly.]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/182/182#msg-182</guid>
      <pubDate>Tue, 20 Feb 2007 14:51:30 +0000</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/161#msg-161</link>
      <author>moderator</author>
      <description><![CDATA[Note: you have a spelling mistake here 
&gt;&gt;register =&gt; 076xxxxxxx:xxxxxxx@freespeach.ie/076xxxxxxx 

! as peachy as the service is]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/161#msg-161</guid>
      <pubDate>Thu, 11 Jan 2007 12:51:31 +0000</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/160#msg-160</link>
      <author>moderator</author>
      <description><![CDATA[1. Check if your Asterisk box registered by logging into freespeech account.
2. Peform a trace on the Asterisk command line and see what is happening

&gt;SIP DEBUG 

( oops... probably not a good idea to post your passsword to the forum )]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/160#msg-160</guid>
      <pubDate>Thu, 11 Jan 2007 12:50:12 +0000</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/159#msg-159</link>
      <author>belgarath</author>
      <description><![CDATA[i solved this]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/159#msg-159</guid>
      <pubDate>Thu, 11 Jan 2007 12:44:12 +0000</pubDate>
    </item>
    <item>
      <title>Quick install</title>
      <link>http://forum.freespeech.ie/read/14/76/76#msg-76</link>
      <author>lzbones</author>
      <description><![CDATA[This might be of interest to those looking for a quick install with asterisk. this is a FULL asterisk system with GUI - should work straight after install. NOTE: this is a complete operating system, it will erase your disks.

http://www.trixbox.org/]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/76/76#msg-76</guid>
      <pubDate>Thu, 15 Jun 2006 17:05:42 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/67#msg-67</link>
      <author>Crime master go go</author>
      <description><![CDATA[yeah...use this for G729 support - http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/

be aware of patent issues ;)

you can download binary and pop into /usr/lib/asterisk/modules/ then configure your channels for G729 in sip.conf]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/67#msg-67</guid>
      <pubDate>Wed, 24 May 2006 13:08:12 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/66#msg-66</link>
      <author>david.marshall</author>
      <description><![CDATA[:-) ok I get it, get it, get it...   ta.

I downloaded a copy of Asterisk@home a couple of months ago.  From what I can see I just burn it to CD and bung it in any old workstation/server that I have lying around and let it run from CD.   Then I'll be able to manage it from anywhere on the web

It looks like a good introduction to VoIP...

... I was going to do it this weekend, but now I'm away working away at the weekend again... but I'll try and do it soon.

Any other hints and or tips for the set-up...?

regards
David]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/66#msg-66</guid>
      <pubDate>Fri, 19 May 2006 19:15:33 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/65#msg-65</link>
      <author>moderator</author>
      <description><![CDATA[A note on this 

Unlike traditional phone lines your freespeech account can handle multiple calls, so  you would only add an aditional line if you needed to be reached on more than one number OR you wanted seperate accounting of call charges.]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/65#msg-65</guid>
      <pubDate>Fri, 19 May 2006 10:26:09 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/64#msg-64</link>
      <author>Asterman</author>
      <description><![CDATA[you would do soemthing like this 

[freespeech1]
type=peer
insecure=very ; otherwise authentication errors
nat=yes
username=076XXXXXXX
fromuser=076XXXXXXX
authuser=076XXXXXXX
fromdomain=freespeech.ie
secret=PASSWD
host=freespeech.ie
context=(your inbound context)
dtmfmode=inband
canreinvite=no

[freespeech2]
type=peer
insecure=very ; otherwise authentication errors
nat=yes
username=076YYYYYYY
fromuser=076YYYYYYY
authuser=076YYYYYYY
fromdomain=freespeech.ie
secret=PASSWD
host=freespeech.ie
context=(your inbound context)
dtmfmode=inband
canreinvite=no]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/64#msg-64</guid>
      <pubDate>Thu, 18 May 2006 21:55:16 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/63#msg-63</link>
      <author>david.marshall</author>
      <description><![CDATA[Brilliant.
What would you do if you wanted more than one line?]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/63#msg-63</guid>
      <pubDate>Thu, 18 May 2006 15:23:04 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/58#msg-58</link>
      <author>dollywobble</author>
      <description><![CDATA[I tried that - works perfectly]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/58#msg-58</guid>
      <pubDate>Sat, 29 Apr 2006 14:24:31 +0100</pubDate>
    </item>
    <item>
      <title>Re: anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/52#msg-52</link>
      <author>moderator</author>
      <description><![CDATA[try this..

[general]
register =&gt; 076XXXXXX:PASSWD@freespeech.ie/076XXXXXX

[freespeech]
type=peer
insecure=very ; otherwise authentication errors
nat=yes
username=076XXXXXXX
fromuser=076XXXXXXX
authuser=076XXXXXXX
fromdomain=freespeech.ie
secret=PASSWD
host=freespeech.ie
context=(your inbound context)
dtmfmode=inband
canreinvite=no]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/52#msg-52</guid>
      <pubDate>Wed, 26 Apr 2006 12:44:06 +0100</pubDate>
    </item>
    <item>
      <title>anbody have config details for Asterisk ?</title>
      <link>http://forum.freespeech.ie/read/14/51/51#msg-51</link>
      <author>lzbones</author>
      <description><![CDATA[I'm going to give setting up an Asterisk server a bash.

Anybody manage to do it ??
Anybody got config settings ??]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/51/51#msg-51</guid>
      <pubDate>Tue, 25 Apr 2006 22:12:50 +0100</pubDate>
    </item>
    <item>
      <title>web link</title>
      <link>http://forum.freespeech.ie/read/14/15/15#msg-15</link>
      <author>moderator</author>
      <description><![CDATA[http://www.asterisk.org/]]></description>
      <category>Asterisk</category>
      <guid isPermaLink="true">http://forum.freespeech.ie/read/14/15/15#msg-15</guid>
      <pubDate>Sun, 23 Apr 2006 22:18:04 +0100</pubDate>
    </item>
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