I have a problem like this, but in this case with a tdm400p, after I finish a call it seems asterisk do not end the call so if i try immediately to make another call says that "....are busy now, try...later"<br><br>
<br>Regards!!!<br><br><br><br><div class="gmail_quote">2008/1/29 <<a href="mailto:sunxiujun26@sina.com">sunxiujun26@sina.com</a>>:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
>On Mon, 2008-01-28 at 18:59 +0800, sunxiujun26 at <a href="http://sina.com" target="_blank">sina.com</a> wrote:<br>>> I used trixbox 1.2.3 with a E1 card(not digium).The call often disconnected,and the echo is great.<br>
>> Recently, I bought a E1 card from digium --TE220B. It can not work on trixbox 1.2.3. The &quot;te200series-user-manual&quot; says the card need asterisk 1.2.20 or newer and zaptel 1.2.20 or newer, so I update trixbox1.2.3(asterisk 1.2.12 and zaptel 1.2.9) to trixbox 2.2 (asterisk 1.2.24 and zaptel 1.2.21).But the disconnected rate increased much,the sip_phone can not be used.<br>
>> I don't know if I chose a wrong release ,or I don't use the card correctly. I want somebody can give me some clews.<br>>> Thank you in advance!<br><br>>Do you have callprogress=yes and/or busydetect=yes in your zapata.conf<br>
file? If so, turn those settings off for your E1 channels. They're the<br>most likely cause for disconnects during a call.<br><br>>--<br>>Jared Smith<br>>Community Relations Manager<br>>Digium, Inc.<br><br>
Jared Smith,<br> I am glad to hear from you .<br> I have set callprogress=no and busydetect=no already.<br><br>best wishes!<br><br> sun xiujun<br><br><br><br>-------------------------------------------------------------------<br>
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