I'm running AAH 2.2 and *1 works from my eyebeam sip phones to do on demand recording.<br>
<br>
You need to set the DIAL_OPTIONS of wW in order to utilize this
feature. lower case w means called person can initiate, upper case
means callee can initiate, I think that is the order.<br>
<br>
They show up as auto-<timestamp>-<src>-<dst>.wav in /var/spool/asterisk/monitor<br>
However, they will NOT show up in ARI, I modified the code to show them
and sent the modification to Dan to implement if he chooses.<br>
<br>
-Blake<br><br><div><span class="gmail_quote">On 12/22/05, <b class="gmail_sendername">Tomislav Parcina</b> <<a href="mailto:tparcina@lama.hr">tparcina@lama.hr</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
In article <<a href="mailto:43909.10.50.164.9.1135254641.squirrel@127.0.0.1">43909.10.50.164.9.1135254641.squirrel@127.0.0.1</a>>,<br><a href="mailto:francesco@fampeeters.com">francesco@fampeeters.com</a> says...<br>
> <<a href="http://www.voip-info.org/wiki-Asterisk+config+features.conf">http://www.voip-info.org/wiki-Asterisk+config+features.conf</a>><br>><br>> BTW:<br>> Please let me know when you've got this working 100%... I keep having
<br>> issues with it! Most notably when dialling OUTBOUND with IAX softphone<br>> (tried borg DIAX and IDEFISK)<br><br>I have tried it with softphones but it didn't work. Now I have three<br>diferent SIP phones and Digium TDM22P card. So I can try diferent sort
<br>of combinations. Hopefully one of them will work :))<br><br><br>--<br><br>Tomislav Parcina<br><a href="mailto:ime.prezime@email.t-com.hr">ime.prezime@email.t-com.hr</a><br><br>_______________________________________________
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