<div dir="ltr">hello yes this is a fresh install <div><br></div><div><div>[trunkgroups]</div><div>trunkgroup => 1,16</div><div>spanmap => 1,1,1</div><div><br></div><div>[channels]</div><div>#include dahdi-channels.conf</div>
<div><br></div><div>context=default</div><div>hidecallerid=no</div><div>callwaiting=yes</div><div>usecallingpres=yes</div><div>callwaitingcallerid=yes</div><div>threewaycalling=yes</div><div>transfer=yes</div><div>canpark=yes</div>
<div>cancallforward=yes</div><div>callreturn=yes</div><div>rxgain=0.0</div><div>txgain=0.0</div><div><br></div><div>group=1</div><div>switchtype=euroisdn</div><div>signalling=pri_cpe</div><div>callgroup=1</div><div>pickupgroup=1</div>
<div>immediate=no</div><div>channel => 1-15,17-31</div><div><br></div><div style>the issue h=just with group 1 can not call via G1</div><div style><br></div><div style>with group 2 theris no problem</div><div><br></div>
<div>group=2</div><div>callgroup=2</div><div>switchtype=qsig</div><div>signalling=pri_net</div><div>callerid=520xxxxxx</div><div>immediate=no</div><div>channel => 32-46,48-52</div></div><div><br></div><div><br></div><div style>
thanks and regards</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/10/21 John Novack <span dir="ltr"><<a href="mailto:jnovack@stromberg-carlson.org" target="_blank">jnovack@stromberg-carlson.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<font size="+1">A VERY OLD and beyond EOF version.<br>
If you MUST, due to some driver issue, use Asterisk 1.4, then
please use 1.4.44<br>
Otherwise I suggest you move to something more current, either
version 1.8.current or beyond.<br>
Also, CLI says 1.4.43, your message says 1.4.32 ???<br>
<br>
Some examination of chan_dahdi and your dialplan would help
someone give you some assistance.<br>
Is this a fresh install, or one that has been working for years?<br>
<br>
What Digium card?<br>
<br>
John Novack<br>
<br>
</font><div><div class="h5">
<div>Salaheddine Elharit wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>i need your help regarding some issue related to the
outband calls</div>
<div><br>
</div>
<div>i have installed asterisk 1.4.32 with dahdi and i have 1
card diguim with 2 ports </div>
<div>when i try to call my phone number all time i receive
message busy number </div>
<div><br>
</div>
<div>this error just with g1.</div>
<div><br>
</div>
<div>with g2 there is no problem i can call without
issue</div>
<div><br>
</div>
<div>can anyone see the CLI and tell me what is the problem</div>
<div><br>
</div>
<div>thanks and regards</div>
<div><br>
</div>
<div>
<div> == Parsing '/etc/asterisk/asterisk.conf': Found</div>
<div> == Parsing '/etc/asterisk/extconfig.conf': Found</div>
<div>
Connected to Asterisk 1.4.43.18495-AheevaCCS-3.2.12
currently running on SRVRADI
O
(pid = 4147)</div>
<div>Verbosity is at least 3</div>
<div> -- Executing [0661049303@agents:1]
Set("SIP/223-00000021", "CALLERID(number)
=520460587") in new stack</div>
<div> -- Executing [0661049303@agents:2]
Dial("SIP/223-00000021", "DAHDI/g1/066104
9303|30") in new stack</div>
<div> -- Requested transfer capability: 0x00 - SPEECH</div>
<div> -- Called g1/0661049303</div>
<div> -- Moving call (DAHDI/3-1) from channel 3 to 2.</div>
<div>[Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9438
pri_fixup_principle: Can't mo
ve
call (DAHDI/3-1) from channel 3 to 2. It is already in use.</div>
<div>[Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9558
pri_find_fixup_principle: Spa
n
1: PRI requested channel 1/2 is not available.</div>
<div> -- Hungup 'DAHDI/3-1'</div>
<div> == Everyone is busy/congested at this time (1:0/0/1)</div>
<div> -- Executing [0661049303@agents:3]
Hangup("SIP/223-00000021", "") in new sta
ck</div>
<div> == Spawn extension (agents, 0661049303, 3) exited
non-zero on 'SIP/223-0000002
1'</div>
<div> -- Executing [h@agents:1] GotoIf("SIP/223-00000021",
"0?3:2") in new stack</div>
<div> -- Goto (agents,h,2)</div>
<div> -- Executing [h@agents:2]
AHEventsProxy("SIP/223-00000021", "MSG_TYPE_TERMIN
ATE_CALL::::1382377407") in new stack</div>
<div> AHEventsProxy: Channel [SIP/223-00000021]. Data
[MSG_TYPE_TERMINATE_CALL::::138
2377407]</div>
<div> -- chan is SIP/223-00000021</div>
<div> AHEventsProxy: Send To CtiServer: socket:[89].
message:[41,1382377407^^^^stcrpb
x^~]</div>
<div> -- Executing [h@agents:3] Hangup("SIP/223-00000021",
"") in new stack</div>
<div> == Spawn extension (agents, h, 3) exited non-zero on
'SIP/223-00000021'</div>
<div> -- SIP/224-00000020 is ringing</div>
<div>SRVRADIO*CLI></div>
<div>Disconnected from Asterisk server</div>
<div>Executing last minute cleanups</div>
<div><br>
</div>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</blockquote>
<br>
</div></div><span class="HOEnZb"><font color="#888888"><pre cols="10000">--
Dog is my Co-pilot</pre>
</font></span></div>
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