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VoipRadioPanel - Revision history
2024-03-28T22:08:24Z
Revision history for this page on the wiki
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https://revspace.nl/index.php?title=VoipRadioPanel&diff=6358&oldid=prev
Maxell at 08:44, 17 April 2015
2015-04-17T08:44:56Z
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<td colspan="2" style="background-color: #fff; color: #222; text-align: center;">← Older revision</td>
<td colspan="2" style="background-color: #fff; color: #222; text-align: center;">Revision as of 08:44, 17 April 2015</td>
</tr><tr><td colspan="2" class="diff-lineno" id="mw-diff-left-l1" >Line 1:</td>
<td colspan="2" class="diff-lineno">Line 1:</td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>= <del class="diffchange diffchange-inline">%NOTOC%</del>Goal =</div></td><td class='diff-marker'>+</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>= Goal =</div></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div> </div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Have max. 6 participants dial on to an asterisk pbx, make each of their voices individually available as a mono output. Each participant should hear the other participants plus an external input from the mixing desk (studio mics / music), but not hear themselves back.</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Have max. 6 participants dial on to an asterisk pbx, make each of their voices individually available as a mono output. Each participant should hear the other participants plus an external input from the mixing desk (studio mics / music), but not hear themselves back.</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Documentation should be provider for common platforms (linux, macosx, ms windows). Pick one voip client on each platform, and document every step from installation, configuration to connecting to the conference pbx and joining.</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Documentation should be provider for common platforms (linux, macosx, ms windows). Pick one voip client on each platform, and document every step from installation, configuration to connecting to the conference pbx and joining.</div></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;">__FORCETOC__</del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>= Todo =</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>= Todo =</div></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Needs a lot of testing in the field:</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Needs a lot of testing in the field:</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td colspan="2" class="diff-lineno" id="mw-diff-left-l17" >Line 17:</td>
<td colspan="2" class="diff-lineno">Line 13:</td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>= Setup =</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>= Setup =</div></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* ESI u46xl sound card (6 out, 4 in, sponsored by Hxx)</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* ESI u46xl sound card (6 out, 4 in, sponsored by Hxx)</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* Mixing desk with at least 6 line-ins, one aux or sub out</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* Mixing desk with at least 6 line-ins, one aux or sub out</div></td></tr>
<tr><td colspan="2" class="diff-lineno" id="mw-diff-left-l25" >Line 25:</td>
<td colspan="2" class="diff-lineno">Line 20:</td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>== ESI u46xl ==</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>== ESI u46xl ==</div></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Linux has crappy support for this device. To initialize the device properly, send the following to the usb mixer with 'amixer -c 1 -s':</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Linux has crappy support for this device. To initialize the device properly, send the following to the usb mixer with 'amixer -c 1 -s':</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td colspan="2" class="diff-lineno" id="mw-diff-left-l1206" >Line 1,206:</td>
<td colspan="2" class="diff-lineno">Line 1,200:</td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>== Known issues ==</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>== Known issues ==</div></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* siptosis - spy output is very choppy, is this for all sip channels?? needs testing, spy works perfect when spying on iax channels</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* siptosis - spy output is very choppy, is this for all sip channels?? needs testing, spy works perfect when spying on iax channels</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* killing siptoskype without hanging up the incoming skype call will result in odd lingering channels in asterisk, messing with your head</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>* killing siptoskype without hanging up the incoming skype call will result in odd lingering channels in asterisk, messing with your head</div></td></tr>
</table>
Maxell
https://revspace.nl/index.php?title=VoipRadioPanel&diff=2680&oldid=prev
Maxell at 17:23, 26 November 2012
2012-11-26T17:23:16Z
<p></p>
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<td colspan="2" style="background-color: #fff; color: #222; text-align: center;">← Older revision</td>
<td colspan="2" style="background-color: #fff; color: #222; text-align: center;">Revision as of 17:23, 26 November 2012</td>
</tr><tr><td colspan="2" class="diff-lineno" id="mw-diff-left-l1" >Line 1:</td>
<td colspan="2" class="diff-lineno">Line 1:</td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div><del style="font-weight: bold; text-decoration: none;"></del></div></td><td colspan="2"> </td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>= %NOTOC%Goal =</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>= %NOTOC%Goal =</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td colspan="2" class="diff-lineno" id="mw-diff-left-l556" >Line 556:</td>
<td colspan="2" class="diff-lineno">Line 554:</td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>== siptosis ==</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>== siptosis ==</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'>−</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #ffe49c; vertical-align: top; white-space: pre-wrap;"><div>Get the gateway software [<del class="diffchange diffchange-inline">httphttp</del>://www.mhspot.com/sts/siptosis_download.php here].</div></td><td class='diff-marker'>+</td><td style="color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #a3d3ff; vertical-align: top; white-space: pre-wrap;"><div>Get the gateway software [<ins class="diffchange diffchange-inline">http</ins>://www.mhspot.com/sts/siptosis_download.php here].</div></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"></td></tr>
<tr><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Some useful documents:</div></td><td class='diff-marker'> </td><td style="background-color: #f8f9fa; color: #222; font-size: 88%; border-style: solid; border-width: 1px 1px 1px 4px; border-radius: 0.33em; border-color: #eaecf0; vertical-align: top; white-space: pre-wrap;"><div>Some useful documents:</div></td></tr>
</table>
Maxell
https://revspace.nl/index.php?title=VoipRadioPanel&diff=271&oldid=prev
10.42.42.9: Created page with " = %NOTOC%Goal = Have max. 6 participants dial on to an asterisk pbx, make each of their voices individually available as a mono output. Each participant should hear the other ..."
2011-10-01T23:26:23Z
<p>Created page with " = %NOTOC%Goal = Have max. 6 participants dial on to an asterisk pbx, make each of their voices individually available as a mono output. Each participant should hear the other ..."</p>
<p><b>New page</b></p><div><br />
<br />
= %NOTOC%Goal =<br />
<br />
Have max. 6 participants dial on to an asterisk pbx, make each of their voices individually available as a mono output. Each participant should hear the other participants plus an external input from the mixing desk (studio mics / music), but not hear themselves back.<br />
<br />
Documentation should be provider for common platforms (linux, macosx, ms windows). Pick one voip client on each platform, and document every step from installation, configuration to connecting to the conference pbx and joining.<br />
<br />
__FORCETOC__<br />
<br />
= Todo =<br />
<br />
Needs a lot of testing in the field:<br />
<br />
* get a bunch of typical users to give up a few hours of their time, explain they will need to be patient and do exactly as we say<br />
* set up a conference bridge with different types of connections: pots, voip, skype to sip gateway<br />
* documentation<br />
* create sample set to guide users in joining the conf<br />
<br />
= Setup =<br />
<br />
* ESI u46xl sound card (6 out, 4 in, sponsored by Hxx)<br />
* Mixing desk with at least 6 line-ins, one aux or sub out<br />
* linphone instances to connect to the pbx<br />
* asterisk with custom extensions and scripts<br />
* siptosis skype gateway<br />
<br />
== ESI u46xl ==<br />
<br />
Linux has crappy support for this device. To initialize the device properly, send the following to the usb mixer with 'amixer -c 1 -s':<br />
<br />
<pre><br />
cset numid=1 on,on<br />
cset numid=2 on<br />
cset numid=3 0,0,0,0<br />
cset numid=4 on,on<br />
cset numid=1 off,off<br />
cset numid=5 off<br />
cset numid=1 on,on<br />
cset numid=7 on,on,on,on<br />
cset numid=8 on<br />
cset numid=6 0,0,0,0<br />
cset numid=9 31,31,31,31,31,31<br />
</pre><br />
<br />
(yes, we turn 1 on, of and on again, needed to properly load 5, which disables monitor)<br />
<br />
This .asoundrc makes the channels individually adressable and creates some fake input/output for linphone:<br />
<br />
<pre><br />
// Copyright (c) Koen Martens, gmc@sonologic.nl<br />
// licensed under GPLv3<br />
<br />
pcm.u46xl_play {<br />
type dmix<br />
ipc_key 2048<br />
slave {<br />
pcm "hw:1,0"<br />
rate 44100<br />
period_time 0<br />
period_size 1024<br />
buffer_size 8192<br />
channels 6<br />
}<br />
bindings {<br />
0 0<br />
1 1<br />
2 2<br />
3 3<br />
4 4<br />
5 5<br />
}<br />
}<br />
<br />
pcm.out1 {<br />
type plug<br />
slave {<br />
pcm "u46xl_play"<br />
channels 6<br />
rate 44100<br />
}<br />
ttable.0.0 1<br />
}<br />
pcm.out2 {<br />
type plug<br />
slave {<br />
pcm "u46xl_play"<br />
channels 6<br />
}<br />
ttable.0.1 1<br />
}<br />
pcm.out3 {<br />
type plug<br />
slave {<br />
pcm "u46xl_play"<br />
channels 6<br />
}<br />
ttable.0.2 1<br />
}<br />
pcm.out4 {<br />
type plug<br />
slave {<br />
pcm "u46xl_play"<br />
channels 6<br />
}<br />
ttable.0.3 1<br />
}<br />
pcm.out5 {<br />
type plug<br />
slave {<br />
pcm "u46xl_play"<br />
channels 6<br />
}<br />
ttable.0.4 1<br />
}<br />
pcm.out6 {<br />
type plug<br />
slave {<br />
pcm "u46xl_play"<br />
channels 6<br />
}<br />
ttable.0.5 1<br />
}<br />
<br />
pcm.u46xl_record {<br />
type dsnoop<br />
ipc_key 1647<br />
slave {<br />
pcm "hw:1,0"<br />
period_size 0<br />
buffer_size 65536<br />
rate 44100<br />
channels 4<br />
}<br />
bindings {<br />
0 0<br />
1 1<br />
}<br />
}<br />
<br />
pcm.in1 {<br />
type plug<br />
ttable.0.0 1.0<br />
slave.pcm pcm.u46xl_record<br />
}<br />
<br />
pcm.in2 {<br />
type plug<br />
ttable.0.1 1.0<br />
slave.pcm pcm.u46xl_record<br />
}<br />
<br />
pcm.in3 {<br />
type plug<br />
ttable.0.2 1.0<br />
slave.pcm pcm.u46xl_record<br />
}<br />
<br />
pcm.in4 {<br />
type plug<br />
ttable.0.3 1.0<br />
slave.pcm pcm.u46xl_record<br />
}<br />
pcm.dummy {<br />
type plug<br />
slave {<br />
pcm default<br />
channels 1<br />
rate 44100<br />
}<br />
ttable.0.0 0<br />
}<br />
<br />
pcm.mynull {<br />
type null<br />
}<br />
<br />
pcm.channel1 {<br />
type asym<br />
playback.pcm "out1"<br />
capture.pcm "dummy"<br />
}<br />
<br />
pcm.channel2 {<br />
type asym<br />
playback.pcm "out2"<br />
capture.pcm "dummy"<br />
}<br />
<br />
pcm.channel3 {<br />
type asym<br />
playback.pcm "out3"<br />
capture.pcm "dummy"<br />
}<br />
<br />
pcm.channel4 {<br />
type asym<br />
playback.pcm "out4"<br />
capture.pcm "dummy"<br />
}<br />
pcm.channel5 {<br />
type asym<br />
playback.pcm "out5"<br />
capture.pcm "dummy"<br />
}<br />
<br />
pcm.channel6 {<br />
type asym<br />
playback.pcm "out6"<br />
capture.pcm "dummy"<br />
}<br />
<br />
pcm.feedback {<br />
type asym<br />
playback.pcm "null"<br />
capture.pcm "in1"<br />
}<br />
<br />
</pre><br />
<br />
== linphone ==<br />
<br />
Use linphonec -c configfile -s 'spyext@voip.host', eg:<br />
<br />
<pre><br />
linphonec -c linphone1.cfg -s '1@pbx.sonologic.net'<br />
</pre><br />
<br />
Here is linphone1.cfg:<br />
<br />
<pre><br />
[sound]<br />
alsadev=channel1<br />
playback_dev_id=ALSA: channel1<br />
ringer_dev_id=ALSA: channel1<br />
capture_dev_id=ALSA: channel1<br />
remote_ring=/usr/share/sounds/linphone/ringback.wav<br />
<br />
[GtkUi]<br />
advanced_ui=1<br />
<br />
[video]<br />
device=V4L2: /dev/video0<br />
size=cif<br />
enabled=0<br />
display=0<br />
capture=0<br />
show_local=0<br />
self_view=0<br />
<br />
[net]<br />
download_bw=0<br />
upload_bw=0<br />
firewall_policy=2<br />
mtu=0<br />
stun_server=pbx.sonologic.net<br />
<br />
[sip]<br />
sip_port=5071<br />
guess_hostname=1<br />
inc_timeout=15<br />
use_info=0<br />
use_rfc2833=0<br />
use_ipv6=0<br />
register_only_when_network_is_up=1<br />
default_proxy=-1<br />
contact=<sip:spy@localhost:5071><br />
<br />
[rtp]<br />
audio_rtp_port=7081<br />
video_rtp_port=9078<br />
audio_jitt_comp=60<br />
video_jitt_comp=0<br />
nortp_timeout=30<br />
<br />
[audio_codec_0]<br />
mime=speex<br />
rate=32000<br />
enabled=1<br />
<br />
[audio_codec_1]<br />
mime=speex<br />
rate=16000<br />
enabled=1<br />
<br />
[audio_codec_2]<br />
mime=speex<br />
rate=8000<br />
enabled=1<br />
<br />
[audio_codec_3]<br />
mime=GSM<br />
rate=8000<br />
enabled=1<br />
<br />
[audio_codec_4]<br />
mime=PCMU<br />
rate=8000<br />
enabled=1<br />
<br />
[audio_codec_5]<br />
mime=PCMA<br />
rate=8000<br />
enabled=1<br />
<br />
[video_codec_0]<br />
mime=MP4V-ES<br />
rate=90000<br />
enabled=1<br />
recv_fmtp=profile-level-id=3<br />
<br />
[video_codec_1]<br />
mime=theora<br />
rate=90000<br />
enabled=1<br />
<br />
[video_codec_2]<br />
mime=H263-1998<br />
rate=90000<br />
enabled=1<br />
recv_fmtp=CIF=1;QCIF=1<br />
<br />
[video_codec_3]<br />
mime=H263<br />
rate=90000<br />
enabled=1<br />
<br />
[video_codec_4]<br />
mime=x-snow<br />
rate=90000<br />
enabled=1<br />
<br />
[auth_info_0]<br />
username=spyuser<br />
userid=spyuser<br />
passwd=secret<br />
realm="asterisk"<br />
</pre><br />
<br />
Be sure to use different alsa devices and different sip/rtp ports for additional spying clients.<br />
<br />
For feedback (audio from the mixing desk sub out to the conference):<br />
<br />
<pre><br />
linphonec -c feedback.cfg -s 'feedback@pbx.sonologic.net'<br />
</pre><br />
<br />
Here is the config:<br />
<br />
<pre><br />
[sound]<br />
alsadev=feedback<br />
playback_dev_id=ALSA: feedback<br />
ringer_dev_id=ALSA: feedback<br />
capture_dev_id=ALSA: feedback<br />
remote_ring=/usr/share/sounds/linphone/ringback.wav<br />
<br />
[GtkUi]<br />
advanced_ui=1<br />
<br />
[video]<br />
device=V4L2: /dev/video0<br />
size=cif<br />
enabled=0<br />
display=0<br />
capture=0<br />
show_local=0<br />
self_view=0<br />
<br />
[net]<br />
download_bw=0<br />
upload_bw=0<br />
firewall_policy=2<br />
mtu=0<br />
stun_server=p��<br />
<br />
[sip]<br />
sip_port=5080<br />
guess_hostname=1<br />
inc_timeout=15<br />
use_info=0<br />
use_rfc2833=0<br />
use_ipv6=0<br />
register_only_when_network_is_up=1<br />
default_proxy=-1<br />
contact=<sip:spy@localhost:5070><br />
<br />
[rtp]<br />
audio_rtp_port=7090<br />
video_rtp_port=9078<br />
audio_jitt_comp=60<br />
video_jitt_comp=0<br />
nortp_timeout=30<br />
<br />
[audio_codec_0]<br />
mime=speex<br />
rate=32000<br />
enabled=1<br />
<br />
[audio_codec_1]<br />
mime=speex<br />
rate=16000<br />
enabled=1<br />
<br />
[audio_codec_2]<br />
mime=speex<br />
rate=8000<br />
enabled=1<br />
<br />
[audio_codec_3]<br />
mime=GSM<br />
rate=8000<br />
enabled=1<br />
<br />
[audio_codec_4]<br />
mime=PCMU<br />
rate=8000<br />
enabled=1<br />
<br />
[audio_codec_5]<br />
mime=PCMA<br />
rate=8000<br />
enabled=1<br />
<br />
[video_codec_0]<br />
mime=MP4V-ES<br />
rate=90000<br />
enabled=1<br />
recv_fmtp=profile-level-id=3<br />
<br />
[video_codec_1]<br />
mime=theora<br />
rate=90000<br />
enabled=1<br />
<br />
[video_codec_2]<br />
mime=H263-1998<br />
rate=90000<br />
enabled=1<br />
recv_fmtp=CIF=1;QCIF=1<br />
<br />
[video_codec_3]<br />
mime=H263<br />
rate=90000<br />
enabled=1<br />
<br />
[video_codec_4]<br />
mime=x-snow<br />
rate=90000<br />
enabled=1<br />
<br />
[auth_info_0]<br />
username=spy<br />
passwd=secret<br />
realm="asterisk"<br />
</pre><br />
<br />
== extensions.ael ==<br />
<br />
You need a sip.conf with a spy user and a feedback user, which both are in context spy.<br />
<br />
You may want to create sip accounts for the panel members. Dump them in context panel.<br />
<br />
You might want to make an extension 'panel' in your public sip context, which goes to panel|s|1.<br />
<br />
<pre><br />
// Copyright (c) 2011, Koen Martens <gmc@sonologic.nl><br />
// licensed under GPLv3<br />
<br />
context panel {<br />
s => {<br />
Answer(500);<br />
BackGround(auth-incorrect);<br />
WaitExten;<br />
};<br />
t => {<br />
Hangup();<br />
};<br />
_XXXX => {<br />
if (${DB_EXISTS(PANEL_ACCOUNT/${EXTEN})}) {<br />
Set(participantno=${DB(PANEL_ACCOUNT/${EXTEN})});<br />
jump s@panel_join;<br />
} else {<br />
Playback(auth-incorrect);<br />
jump s;<br />
}<br />
};<br />
h => {<br />
NoOp(${DB_DELETE(PANEL_CHANNELS/${participantno})});<br />
};<br />
};<br />
<br />
context panel_join {<br />
s => {<br />
// expects participantno to be set<br />
if (${DB_EXISTS(PANEL_CHANNELS/${participantno})}) {<br />
// Warn about participant already existing<br />
}<br />
Set(DB(PANEL_CHANNELS/${participantno})=${CHANNEL});<br />
// ConfBridge(0,1q);<br />
MeetMe(0,1q);<br />
}<br />
}<br />
<br />
context spy {<br />
_X => {<br />
Answer(500);<br />
while (1) {<br />
if (${DB_EXISTS(PANEL_CHANNELS/${EXTEN})}) {<br />
Set(channeltospy=${DB(PANEL_CHANNELS/${EXTEN})});<br />
//ChanSpy(${channeltospy},oqXS);<br />
ChanSpy(${channeltospy},oXSv(4));<br />
}<br />
Wait(5);<br />
}<br />
};<br />
_feedback => {<br />
Answer(500);<br />
// ConfBridge(0,1qs);<br />
MeetMe(0,1q);<br />
}<br />
echo => {<br />
Answer(500);<br />
Echo();<br />
Hangup();<br />
}<br />
};<br />
<br />
context moderator {<br />
// Moderator for conference<br />
1337 => {<br />
Answer(2000);<br />
// Warn about being joined as a moderator<br />
// ConfBridge(0,1qsa);<br />
MeetMe(0,1qsa);<br />
};<br />
// Dial out<br />
_0X. => {<br />
Set(outboundno=${EXTEN:1});<br />
jump s@outboundtie;<br />
}<br />
};<br />
<br />
context outboundtie {<br />
s => {<br />
Dial(SIP/moderator,60,T);<br />
NoOp();<br />
};<br />
_X => {<br />
Set(participantno=${EXTEN});<br />
jump s@panel_join;<br />
}<br />
}<br />
</pre><br />
<br />
== siptosis ==<br />
<br />
Get the gateway software [httphttp://www.mhspot.com/sts/siptosis_download.php here].<br />
<br />
Some useful documents:<br />
* http://www.mhspot.com/sts/siptosis_config_issues.html<br />
* http://www.mhspot.com/sts/siptosis_skype_trunk_howto.html<br />
<br />
I am not pretending I know exactly what is going on here, but something like this should do the trick (supposing there is a sip account panel2 on your pbx):<br />
<br />
<pre><br />
# siptosis.cfg<br />
debug_level=3<br />
<br />
# SipToSis configuration file<br />
<br />
# ___________________________________________<br />
<br />
#<br />
<br />
<br />
<br />
#how opften in minutes to check for configuration changes (0=disable)<br />
<br />
configWatchInterval=0<br />
<br />
<br />
<br />
#Set to log_debug.properties for full debugging<br />
<br />
logConfigFile=log.properties<br />
<br />
<br />
<br />
#Files containing Authorization rules<br />
<br />
siptoskypeauthfile=SipToSkypeAuth.props<br />
<br />
skypetosipauthfile=SkypeToSipAuth.props<br />
<br />
<br />
<br />
#Files containing dialing transforms<br />
<br />
SkypeOutDialingRulesFile=SkypeOutDialingRules.props<br />
<br />
SipOutDialingRulesFile=SipOutDialingRules.props<br />
<br />
<br />
<br />
#location of ua.jar file<br />
<br />
ua_jar=ua.jar<br />
<br />
<br />
<br />
#increase audio threads processing priority (0-2) 0=normal<br />
<br />
audioPriorityIncrease=0<br />
<br />
<br />
<br />
#jitter enable and size (0=disable,1=small,2=medium,3=large,4=extra large)<br />
<br />
jitterLevel=2<br />
<br />
<br />
<br />
#Set to skype_connect=no to disable connection to skype client - for testing purposes<br />
<br />
skype_connect=yes<br />
<br />
<br />
<br />
#Following ports are used by skype to transfer audio to/from siptosis <br />
<br />
# - use any unused ports - uses 2 ports per connection<br />
<br />
skype_audioportbase=64432<br />
<br />
<br />
<br />
<br />
<br />
#Set to yes to enable skype DTMF support - uses more cpu<br />
<br />
enableSkypeDtmfDetector=no<br />
<br />
SkypeDtmfDetectorHitThreshold=90<br />
<br />
SkypeDtmfDetectorSilenceThreshold=6<br />
<br />
<br />
<br />
#Set to yes to regenerate SIP DTMF to Skype<br />
<br />
sendSipDtmfToSkype=yes<br />
<br />
<br />
<br />
#Set to yes to regenerate Skype DTMF to Sip<br />
<br />
sendSkypeDtmfToSip=yes<br />
<br />
<br />
<br />
#If using inband detectors, no to detect dtmf only during authentication (saves cpu)<br />
<br />
inbandFullTimeDtmfDetection=no<br />
<br />
<br />
<br />
<br />
<br />
#special mode if using skype client manually and an outbound skype call is made, it will attempt a sip call and link the two<br />
<br />
JoinManualSkypeOutboundCallToSip=no<br />
<br />
<br />
<br />
#refuse,voicemail,ignore,transferto:skypeid <br />
<br />
# If you are using a PBX with multiple clients/ids you probably want to use ignore<br />
<br />
# or possibly transferto:nextskypeid to the next skypeclient in the chain<br />
<br />
# then refuse on the last one<br />
<br />
SkypeInboundAllChannelsBusyAction=refuse<br />
<br />
<br />
<br />
#If an incoming skype call and sip destination is not available for any reason, what to do with skype call. <br />
<br />
#Allowed options: ring or refuse - ring allows the skype client to continue ringing and be answered manually.<br />
<br />
SkypeInboundSipDestUnavailableAction=refuse<br />
<br />
<br />
<br />
<br />
<br />
#busy,transferto:sipurl <br />
<br />
# If you want multiple outbound channels (AsteriskWin32 does not like this)<br />
<br />
# use transferto:sipurl to the next SipToSis channel in the chain then busy on the last one<br />
<br />
SipInboundAllChannelsBusyAction=busy<br />
<br />
<br />
<br />
<br />
<br />
#enable if skype client can support multiple active calls at same time - trying to lie here won't work<br />
<br />
#I have yet to find a client that can do this<br />
<br />
skypeclientsupportsmulticalls=no<br />
<br />
<br />
<br />
#For an ATA/SIP Phone set to 2 - this allows two total calls - one will be on hold.<br />
<br />
#For a PBX - depending on if the skype client supports multi calls - if not set it to 1<br />
<br />
# otherwise set based on your hardware/bandwidth limitations<br />
<br />
concurrentcalllimit=2<br />
<br />
<br />
<br />
#specify in 5 minute increments, 0=disable auto shutdown - siptosis will auto shutdown in x minutes when idle<br />
<br />
autoShutdownMinutes=0<br />
<br />
<br />
<br />
#Seconds caller has to enter the pin number<br />
<br />
pintimeout=8<br />
<br />
#Number of pin entry attempts before auto hangup<br />
<br />
pinretrylimit=3<br />
<br />
<br />
<br />
#Seconds caller has to enter the destination number<br />
<br />
destinationtimeout=12<br />
<br />
#Number of destination entry attempts before auto hangup<br />
<br />
destinationretrylimit=3<br />
<br />
<br />
<br />
#SIP authorization system recordings - make your own if you like (wav 16k 16bit mono).<br />
<br />
pinFile=clips/enterPin.wav<br />
<br />
destinationFile=clips/enterDest.wav<br />
<br />
dialingFile=clips/dialing.wav<br />
<br />
invalidPinFile=clips/invalidPin.wav<br />
<br />
invalidDestFile=clips/invalidDest.wav<br />
<br />
<br />
<br />
#Skype authorization system recordings - make your own if you like (wav 16k 16bit mono).<br />
<br />
skypePinFile=clips/enterPin.wav<br />
<br />
skypeDestinationFile=clips/enterDest.wav<br />
<br />
skypeDialingFile=clips/dialing.wav<br />
<br />
skypeInvalidPinFile=clips/invalidPin.wav<br />
<br />
skypeInvalidDestFile=clips/invalidDest.wav<br />
<br />
<br />
<br />
#Used for Skypeout only - transmit skype feedback sound during PSTN call attempt<br />
<br />
handleEarlyMedia=yes<br />
<br />
<br />
<br />
#Send Skype IM when calling skype users - not used for skypeout<br />
<br />
sendSkypeIM=no<br />
<br />
skypeimmessage=You are about to receive a Skype Voice call from [callerid].<br />
<br />
#delay between the IM and the actual skype call in seconds.<br />
<br />
sendSkypeImDelay=2<br />
<br />
<br />
<br />
#transport_protocols=udp tcp<br />
<br />
transport_protocols=udp<br />
<br />
via_addr=10.1.3.20<br />
<br />
host_ifaddr=ALL-INTERFACES<br />
<br />
enableNatTranslate=yes <br />
<br />
#outbound_proxy=127.0.0.2:5060<br />
<br />
<br />
<br />
#Sample AUTO config with NO registration<br />
<br />
# username and password not important in this mode<br />
<br />
#Set to available port to transport SIP messages on siptosis computer<br />
<br />
#host_port=5070<br />
<br />
#username=skypests<br />
<br />
#passwd=unimportantpassword<br />
<br />
# --- end of NO registration example ---<br />
<br />
<br />
<br />
#Sample config with NO registration - use if above auto config doesn't work - change 127.0.0.1 to ip address of computer running siptosis<br />
<br />
# username and password not important in this mode<br />
<br />
#Set to available port to transport SIP messages on siptosis computer<br />
<br />
#host_port=5070<br />
<br />
#contact_url=sip:skypests@127.0.0.1:5070<br />
<br />
#from_url="skypests" <sip:5611111111@127.0.0.1:5070><br />
<br />
#username=skypests<br />
<br />
#passwd=unimportantpassword<br />
<br />
#realm=127.0.0.1<br />
<br />
# --- end of NO registration example ---<br />
<br />
<br />
<br />
#Sample config WITH registration to GizmoProject - comment out NO registration info above first and uncomment the following<br />
<br />
#contact_url=sip:studiorevspace@127.0.0.1:5070<br />
<br />
#from_url="panel2" <sip:panel2@pbx.sonologic.net:5060><br />
<br />
#username=panel2<br />
<br />
#passwd=secret<br />
#realm=asterisk<br />
<br />
#expires=120<br />
<br />
#minregrenewtime=60<br />
<br />
#regfailretrytime=15<br />
<br />
#do_register=yes<br />
<br />
#outbound_proxy=pbx.sonologic.net:5060<br />
<br />
# --- end of WITH registration example ---<br />
<br />
<br />
<br />
<br />
<br />
#sample FWD reg example - note the outbound proxy<br />
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#host_port=5070<br />
<br />
#contact_url=sip:8?????@SipToSisIpAddress:SipToSisHostPort<br />
<br />
#from_url="8?????" <sip:8?????@fwd.pulver.com:5060><br />
<br />
#username=8?????<br />
<br />
#realm=fwd.pulver.com<br />
<br />
#passwd=????<br />
<br />
#expires=240<br />
<br />
#do_register=yes<br />
<br />
#minregrenewtime=120<br />
<br />
#regfailretrytime=15<br />
<br />
#outbound_proxy=fwdnat2.pulver.com:5060<br />
<br />
# --- end of FWD registration example ---<br />
<br />
<br />
<br />
<br />
<br />
#sample pennytel reg example<br />
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#host_port=5070<br />
<br />
#contact_url=sip:8?????@SipToSisIpAddress:SipToSisHostPort<br />
<br />
#from_url="8?????" <sip:8?????@sip.pennytel.com:5060><br />
<br />
#username=8?????<br />
<br />
#realm=sip.pennytel.com<br />
<br />
#passwd=123456<br />
<br />
#expires=320<br />
<br />
#do_register=yes<br />
<br />
#minregrenewtime=120<br />
<br />
#regfailretrytime=15<br />
<br />
# --- end of pennytel registration example ---<br />
<br />
<br />
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<br />
<br />
#Sample Asterisk registration example - comment out NO registration info above first and uncomment the following.<br />
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#host_port=5070<br />
<br />
#contact_url=sip:skypetestuser@SipToSisIpAddress:SipToSisHostPort<br />
<br />
#from_url="skypetestuser" <sip:skypetestuser@asteriSkipAddress:asteriskHostPort><br />
<br />
#username=skypetestuser<br />
<br />
#realm=asterisk<br />
<br />
#passwd=skypetest<br />
<br />
#expires=3600<br />
<br />
#do_register=yes<br />
<br />
#minregrenewtime=120<br />
<br />
#regfailretrytime=15<br />
<br />
# --- end of Asterisk Reg example ---<br />
<br />
<br />
<br />
#Sample Asterisk registration example - comment out NO registration info above first and uncomment the following.<br />
<br />
host_port=5070<br />
<br />
contact_url=sip:studiorevspace@10.1.3.20:5070<br />
<br />
from_url="panel2" <sip:panel2@pbx.sonologic.net:5060><br />
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username=panel2<br />
<br />
realm=asterisk<br />
<br />
passwd=geheim<br />
expires=3600<br />
<br />
do_register=yes<br />
<br />
minregrenewtime=120<br />
<br />
regfailretrytime=15<br />
<br />
# --- end of Asterisk Reg example ---<br />
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<br />
<br />
#Sample FreeSwitch regisration - can't use 5070 like others<br />
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#host_port=5077<br />
<br />
#contact_url=sip:skypetestuser@SipToSisIpAddress:SipToSisHostPort<br />
<br />
#from_url="skypetestuser" <sip:skypetestuser@freeswitchIPAddress:freeswitchHostPort><br />
<br />
#username=skypetestuser<br />
<br />
#realm=freeswitchDeterminedRealm<br />
<br />
#passwd=skypetest<br />
<br />
#expires=3600<br />
<br />
#do_register=yes<br />
<br />
#minregrenewtime=160<br />
<br />
#regfailretrytime=15<br />
<br />
# --- end of FreeSwitch Reg example ---<br />
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<br />
<br />
<br />
<br />
<br />
<br />
#do_unregister=yes<br />
<br />
#do_unregister_all=yes<br />
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<br />
<br />
#keepalive_time - set to zero to disable keep alives<br />
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keepalive_time=45000<br />
<br />
<br />
<br />
audio=yes<br />
<br />
#following is the SIP RTP port base - use an even port number<br />
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audio_port=63200<br />
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#auto hangup after no rtp packets received for ? seconds<br />
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noRtpReceivedAutoHangupSeconds=30<br />
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<br />
<br />
#only PCMU,PCMA,GSM (jmf lib),GSMTRI (tritonus libs) codecs currently supported - first one is the preferred codec<br />
<br />
#You can append RW to the codec name to disable sample averaging - use need to use filter with those.<br />
<br />
#Speex doesn't not work well at all - mass cpu usage<br />
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audio_codec=PCMU,PCMA,GSMTRI,ILBC,SPEEX<br />
<br />
#PCMU/PCMA allow 160,240,320 - GSM allows 160 - ILBC allows 240, Speex allows 160, Speex16k allows 320 - need one size for each codec specified <br />
<br />
audio_frame_size=240,240,160,240,160<br />
<br />
#if using dynamic payload types (speex and ilbc) you must specify the payload number (asterisk uses 98,97), if not you can remove this parameter<br />
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audio_avp=-1,-1,-1,98,97<br />
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<br />
<br />
#Audio volume gains - 1 for each codec - (decimal number) 1=flat no gain, higher=louder, too high will clip or distort<br />
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#volume sip->skype<br />
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skype_audiooutgain=1,1,1,1,1<br />
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#volume skype->sip<br />
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skype_audioingain=1.5,1.5,1.5,1.5,1.5<br />
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<br />
#Filter skype audio before being downsampled and sent to SIP device. <br />
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#No Filtering<br />
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FilterParams=NONE<br />
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#RC lowpass filter<br />
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#RC,delay time (lower lowers cutoff),RC constant (higher lowers cutoff) - 50,40 is a good starting point <br />
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#FilterParams=RC,50,40<br />
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#FIR filter<br />
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#FIR,Order (higher sharper cutoff and more cpu),window type (RECTANGULAR,HANNING,HAMMING,BLACKMAN),filter type (LP,HP,BP),minFreq,maxFreq<br />
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#FilterParams=FIR,100,HANNING,LP,0,3200<br />
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#FilterParams=FIR,100,HANNING,HP,300,3200<br />
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#FilterParams=FIR,100,HANNING,BP,300,3200<br />
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<br />
<br />
#If yes, will send RTP packets to address received from the otherside<br />
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# instead of what was received in the session descriptor. <br />
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# This may help with one way audio problems. <br />
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enableSendRTPtoReceivedAddress=yes<br />
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#works with above setting - sending of rtpPackets can be redirected until receiving this number of packets. After that the address is locked.<br />
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lockRtpSendAddressAfterPackets=1<br />
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<br />
<br />
#Set to -1 to disable rfc2833 some providers use 96 most use 101<br />
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dtmf2833payloadtype=101 <br />
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<br />
<br />
#Use these for SIP INFO msg support - first is the most common type<br />
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#dtmfinfotype=application/dtmf-relay<br />
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#dtmfinfotype=application/dtmf<br />
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<br />
<br />
#Use only if rfc2833 and INFO are not supported - uses more cpu<br />
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enableSIPInbandDtmfDetector=no<br />
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SipDtmfDetectorHitThreshold=30<br />
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SipDtmfDetectorSilenceThreshold=6<br />
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<br />
<br />
<br />
<br />
#params to control sip response address handling<br />
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useViaRport=yes<br />
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useViaReceived=yes<br />
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#send all responses using outbound proxy - outbound proxy must be set up<br />
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sendResponseUsingOutboundProxy=no<br />
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<br />
<br />
#sip response for any uncovered reason<br />
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baseFailureResponse=403<br />
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#sip response if remote skype user refused call<br />
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skypeRefusedResponse=603<br />
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#sip response if skype call failed, invalid skype user, or no skype credit<br />
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skypeFailedResponse=404<br />
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#sip response if skype returned unplaced status<br />
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skypeUnPlacedResponse=408<br />
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#sip response if called party is busy<br />
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skypeBusyResponse=600<br />
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<br />
<br />
#network buffers for skype api audio transport (0=leave at OS default)<br />
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TcpRxBufferSize=8192<br />
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TcpTxBufferSize=8192<br />
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#network buffers for RTP audio transport (0=leave at OS default)<br />
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RtpRxBufferSize=8192<br />
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RtpTxBufferSize=8192<br />
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<br />
<br />
#*** register server settings below *** not required if registration is not needed for phone<br />
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#set to yes to turn on server registrar or no to disable<br />
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is_registrar=yes<br />
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#set to yes to allow register of users not already in registar database (users.db)<br />
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register_new_users=yes<br />
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#set to domains of server - see mjsip doc<br />
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#domain_names=192.168.0.4 somedomain.com<br />
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allowMultiContactsPerUser=no<br />
</pre><br />
<br />
And you will need something like this in your SkypeToSipAuth.props:<br />
<br />
<pre><br />
*,sip:panel@pbx.sonologic.net:5060<br />
</pre><br />
<br />
Somehow it seems to dial the sip uri without registering, so panel must be in the public incoming SIP context.<br />
<br />
== Known issues ==<br />
<br />
* siptosis - spy output is very choppy, is this for all sip channels?? needs testing, spy works perfect when spying on iax channels<br />
* killing siptoskype without hanging up the incoming skype call will result in odd lingering channels in asterisk, messing with your head</div>
10.42.42.9