<div dir="ltr">Thanks, Steve,<br><br>That's what I am unsure of. I don't know how to limit 1 call per trunk. If that's an easy thing to setup, I'd love to see it.<br><br><div class="gmail_quote">On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.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">Oh, I thought you had logic to count the calls on the trunk. You should limit each trunk to one call. This is the primary reason besides the email that basically said that customer support structure has been changed and anything beyond the Demarc would not be supported, I the Demarc is simply their boxen, so unless it is on their side, you will not get any helpful support from Bandwidth, plus they CCed over 500 people by address instead of setting up a group. <a href="http://www.bandwidth.com/content/support/?page=standardSupport" target="_blank">http://www.bandwidth.com/content/support/?page=standardSupport</a><br>
<br>I am with Junction and while a trunk is not "unlimited" as far as price for usage, the amount of trunks is unlimited (or at least as unlimited as it can be since nothing is really unlimited). They asked that I try not to go over one call per second for any real duration, and that I not hammer one LATA do to limited interconnects. <br>
<br>The other thing was Junctions was very easy to sign up with, great support, and configuration was a breeze.<br><br>As for Bandwidth, I think they are solid but due to recent changes and the fact that you must pay per channel, as well as the setup process, I decided they were not for me.<br>
<br>I will take a second look at your sip.conf and extensions.conf later to see if something jumps out at me. I suspect since you are setting up two separate trunks with Bandwidth, you need to limit each trunk to one call, rather than two.<br>
<br>Thanks,<br><font color="#888888">Steve Totaro</font><div><div></div><div class="Wj3C7c"><br><br><br><br><div class="gmail_quote">On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen <span dir="ltr"><<a href="mailto:kurt.knudsen@gmail.com" target="_blank">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">externip messes up DTMF detection, and by messes up I mean it doesn't detect it at all. Setting nat=yes or nat=no didn't make a difference either.<br>
<br>When the trunks are in use, the calls are fine, no dropped audio. It only happens when a 3rd call is made and there's no trunk available.<br>
<br>Thanks :)<div><div></div><div><br><br><div class="gmail_quote">On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.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">You need to configure your box for nat settings, externip and other settings in sip.conf and set nat=yes instead of nat=no.<br><br>One way audio is almost always a NAT issue and those are two glaring things that would cause problems.<br>
<br>Thanks,<br><font color="#888888">Steve Totaro</font><div><div></div><div><br><br><div class="gmail_quote">On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen <span dir="ltr"><<a href="mailto:kurt.knudsen@gmail.com" target="_blank">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">Hi Steve,<br><br>It's behind a NAT/Firewall but SIP translation is enabled and removing it from behind the firewall did nothing, it still dropped calls. The calls connect and everything works, but it dies when all trunks are in use and someone else tries to call out. It seems like even though both channels are in use, it tries to connect to the 2nd trunk and thus kills the audio. Nothing strange came up in Wireshark or the firewall logs.<br>
<br>Thanks.<br><br><div class="gmail_quote"><div><div></div><div>On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div>
<div dir="ltr"><div><div></div><div><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" target="_blank">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></div></div>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>
<br></div></div></blockquote></div></div></blockquote></div><br></div></div></div></blockquote></div></div></div></div></blockquote></div><br>
</div></div></div>
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