I've set at Protocol Management >> FXO Settings >> Dialing Mode ==> One Stage and everything is fine now<br><br>Thank you very much John,<br><br>EDA<br><br><div class="gmail_quote">On Wed, Dec 2, 2009 at 1:43 PM, John Balogh <span dir="ltr"><<a href="mailto:JDB@psu.edu">JDB@psu.edu</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-US">
<div><div class="im">
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">> I want to do single-stage dialing. I've just realized I </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">> have the two-stage running now (I get dial tone and then,</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">> when i introduce the number, the call get through).</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
</div><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Right. </span></p><div class="im">
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">According to the SIP User's Manual</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf</span></p>
</div><p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">page 67/294</span></p><div class="im">
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">"</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Enable Digit Delivery to Tel [EnableDigitDelivery]</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> Disable [0] = Disabled (default).</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> Enable [1] = Enable Digit Delivery feature for MediaPack/FXO
& FXS.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">The digit delivery feature enables sending of DTMF digits to the
gateway’s port after the line is offhooked (FXS) or seized (FXO). For
IP->Tel calls, after the line is offhooked / seized, the MediaPack plays the
DTMF digits (of the called number) towards the phone line.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">[...]</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">To use this feature with FXO gateways, configure the gateway to
work in one</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">stage dialing mode.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">"</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">You probably need to set the above.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">The FXO parameter (from page 107/294):</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">"</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Dialing Mode [IsTwoStageDial]</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> One Stage [0] = One-stage dialing.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> Two Stage [1] = Two-stage dialing (default).</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Used for IP->FXO gateways calls.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">If two-stage dialing is enabled, then the FXO gateway seizes one
of the PSTN/PBX lines without performing any dial, the remote user is connected
over IP to PSTN/PBX, and all further signaling (dialing and Call Progress
Tones) is performed directly with the PBX without the gateway’s intervention.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">If one-stage dialing is enabled, then the FXO gateway seizes one
of the available lines (according to Channel Select Mode parameter), and dials
the destination phone number received in INVITE message. Use the ‘Waiting For
Dial Tone’ parameter to specify whether the dialing should come after detection
of dial tone, or immediately after seizing of the line.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">"</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">So you probably need to clear that parameter (it is not
configured in your .INI file now, so you need to add it, or change the web
interface drop-down control).</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Let us know if this helps.</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">JDB</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
</div><div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b>On Behalf Of </b>Daniel - Asterisk<div class="im"><br>
<b>Sent:</b> Wednesday, December 02, 2009 12:33 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Help configuring Audiocodes MP-104 FXO</div></span></p>
</div>
<p class="MsoNormal"> </p>
<p class="MsoNormal" style="margin-bottom: 12pt;">Hi list,</p><div><div></div><div class="h5"><br>
<br>
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.<br>
<br>
I very appreciate your advices.<br>
<br>
Command line results and SIPconfigurations follows:<br>
<br>
<b>CLI></b><br>
-- Executing [7991696900@total:1]
Playback("SIP/101-09dd8918", "beep") in new stack<br>
-- <SIP/101-09dd8918> Playing 'beep' (language 'es')<br>
-- Executing [7991696900@total:4] Dial("SIP/101-09dd8918",
"SIP/201/991696900") in new stack<br>
-- Called 201/991696900<br>
-- SIP/201-09ddc890 answered SIP/101-09dd8918<br>
<br>
<br>
<b>sip.conf</b><br>
[201]<br>
secret = ****<br>
callerid = Mobile_01 <201><br>
type = friend<br>
host = dynamic<br>
context = total<br>
dtmfmode=rfc2833<br>
qualify = yes<br>
call-limit=5<br>
disallow = all<br>
allow = gsm<br>
allow = ulaw<br>
allow = alaw<br>
allow = g729</div></div>
</div>
</div>
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