<div dir="ltr"><br><br><div class="gmail_quote">On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <span dir="ltr"><<a href="mailto:kurt.knudsen@gmail.com">kurt.knudsen@gmail.com</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 dir="ltr"><div><div dir="ltr">
<p><span style="font-size: 10pt; font-family: Arial;">Hello,</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">We have 2
SIP trunks from Bandwidth.com and if both are in use and someone tries to dial
out, they cause another call to get one-way audio (the caller hears us, we
cannot hear them). This happens 100% of the time and Bandwidth.com doesn't
offer any support. I don't see any setting that tells Asterisk that there are 2
channels available from Bandwidth.com's IP. I'm currently using, or attempting
to use, groups to solve this problem, but sometimes it works, sometimes it
doesn't. It breaks when a call goes out on a Queue, because it seems to add
each phone to the group, which breaks my GotoIf() statement. Here's some
relevant information:</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">Users.conf
(added by Asterisk-GUI)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[trunk_2]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">provider =
Bandwidth (SIP)<span> </span>; GUI metadata</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">context =
DID_trunk_2</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">hasexten =
no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">hasiax = no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">hassip =
yes</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">host =
<a href="http://216.82.224.202/" target="_blank">216.82.224.202</a></span></p>
<p><span style="font-size: 10pt; font-family: Arial;">registeriax
= no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">registersip
= no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">usecallerid
= yes</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">nat = no
;Testing</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">trunkname =
Bandwidth.com (Sip)<span> </span>; GUI metadata</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">username =</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">secret =</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">disallow =
all</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">allow =
ulaw,alaw,g726</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">sip.conf</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[general]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">context =
frombandwidth</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;other
variables, etc.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;Added
according to Bandwidth.com's wiki entry. Changed to inband because we were
having DTMF issues.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[bandwidth.com_inbound]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">host=<a href="http://216.82.224.202/" target="_blank">216.82.224.202</a></span></p>
<p><span style="font-size: 10pt; font-family: Arial;">port=5060</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">type=peer</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">disallow=all</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">allow=ulaw</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">dtmfmode=inband</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">canreinvite=no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">reinvite=no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">context=frombandwidth</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">nat=no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[bandwidth.com_outbound]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">host=<a href="http://216.82.224.202/" target="_blank">216.82.224.202</a></span></p>
<p><span style="font-size: 10pt; font-family: Arial;">port=5060</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">type=peer</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">disallow=all</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">allow=ulaw</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">dtmfmode=rfc2833</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">nat=no</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">fromuser=11234567890</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">extensions.conf</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[globals]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;…irrelevant
stuff</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">trunk_1 =
Dahdi/g1</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">trunk_2 =
SIP/trunk_2</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">OUT_2 =
SIP/bandwidth.com_outbound</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;Took out
the Set(GROUP()) because I moved it elsewhere to try and fix it added all the
phones when Asterisk calls agents on a Queue.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[frombandwidth]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;exten =
_+1.,1,Set(GROUP()=SIPGROUP)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_+1.,n,Set(DID=${EXTEN:2})</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_+1.,n,Set(CALLERID(num)=${CALLERID(num):2})</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_+1.,n,Goto(DID_trunk_2,${DID},1)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;What we
use to dialout. Try SIP trunks first, then Dahdi trunk as backup.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;This is
where it breaks. I tried to make it so there can't be more than 2 calls on SIP
channels at once.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;Since it
counts the phone as a channel, and adds it to the group, I had to use 4.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[internalphones]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_1NXXNXXXXXX,1,Set(GROUP()=SIPGROUP)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_1NXXNXXXXXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} >= 4]?100)<span> </span>;If the group has 2 or more calls, do not
dial.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten = _1NXXNXXXXXX,n,NoOp(1NCount
= ${GROUP_COUNT(SIPGROUP)})</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_1NXXNXXXXXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_1NXXNXXXXXX,100,Playback(all-circuits-busy-now)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_1NXXNXXXXXX,101,congestion()</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_1NXXNXXXXXX,102,busy()</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">;This is
where incoming calls go to if I'm awake.</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">[DID_trunk_2_timeinterval_Awake]</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_NXXNXXXXXX,1,Set(GROUP()=SIPGROUP)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_NXXNXXXXXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_NXXNXXXXXX,n,Set(CALLERID(num)=1${CALLERID(num)})</span></p>
<p><span style="font-size: 10pt; font-family: Arial;">exten =
_NXXNXXXXXX,n,Goto(voicemenu-custom-1|s|1)</span></p>
<p><span style="font-size: 10pt; font-family: Arial;"> </span></p>
<p><span style="font-size: 10pt; font-family: Arial;">Thanks.</span></p>
</div>
</div></div>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank"></a></blockquote></div><br>Is your Asterisk box on a public IP or behind a NAT/Firewall?<br clear="all"><br>-- <br>Thanks,<br>Steve Totaro <br>
+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br>
</div>