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<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Tried using GROUP()?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">When a call comes in or goes out:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Exten => XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Exten => XXX,n,GotoIf($[${GROUP_COUNT(outgoing/incoming@bdwi_out_1)}] > 1?fail)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Exten => XXX,n,Dial(…)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Exten => XXX(fail),1,Congestion();<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Exten => XXX(fail),n,Hangup();<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Obviously choose outgoing or incoming, if you want to track both you can just use $MATH() to add them together.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Or some other math logic to check the result.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">On incoming Set(DIALSTATUS=CHANUNAVAIL) and it’ll ring busy to bandwidth(or out of service, you can tweak this).<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b>On Behalf Of </b>Kurt Knudsen<br>
<b>Sent:</b> Monday, October 20, 2008 10:46 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">Any updates? It still seems to happen, though not as often as it used to. We're using Polycom 320 phones, if that makes a difference, though we did do it with X-Lite as well.<o:p></o:p></p>
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<p class="MsoNormal">On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen <<a href="mailto:kurt.knudsen@gmail.com">kurt.knudsen@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">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.<o:p></o:p></p>
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<p class="MsoNormal">On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro <<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">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>
<span style="color:#888888">Steve Totaro</span><o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
<br>
<br>
<o:p></o:p></p>
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<p class="MsoNormal">On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen <<a href="mailto:kurt.knudsen@gmail.com" target="_blank">kurt.knudsen@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">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 :)<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro <<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">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>
<span style="color:#888888">Steve Totaro</span><o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen <<a href="mailto:kurt.knudsen@gmail.com" target="_blank">kurt.knudsen@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt">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.<o:p></o:p></p>
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<p class="MsoNormal">On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro <<a href="mailto:stotaro@totarotechnologies.com" target="_blank">stotaro@totarotechnologies.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <<a href="mailto:kurt.knudsen@gmail.com" target="_blank">kurt.knudsen@gmail.com</a>> wrote:<o:p></o:p></p>
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<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">Hello,</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">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><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">Users.conf (added by Asterisk-GUI)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[trunk_2]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">provider = Bandwidth (SIP) ; GUI metadata</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">context = DID_trunk_2</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">hasexten = no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">hasiax = no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">hassip = yes</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">host = <a href="http://216.82.224.202/" target="_blank">
216.82.224.202</a></span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">registeriax = no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">registersip = no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">usecallerid = yes</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">nat = no ;Testing</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">trunkname = Bandwidth.com (Sip) ; GUI metadata</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">username =</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">secret =</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">disallow = all</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">allow = ulaw,alaw,g726</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">sip.conf</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[general]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">context = frombandwidth</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;other variables, etc.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;Added according to Bandwidth.com's wiki entry. Changed to inband because we were having DTMF issues.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[bandwidth.com_inbound]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">host=<a href="http://216.82.224.202/" target="_blank">216.82.224.202</a></span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">port=5060</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">type=peer</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">disallow=all</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">allow=ulaw</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">dtmfmode=inband</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">canreinvite=no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">reinvite=no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">context=frombandwidth</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">nat=no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[bandwidth.com_outbound]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">host=<a href="http://216.82.224.202/" target="_blank">216.82.224.202</a></span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">port=5060</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">type=peer</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">disallow=all</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">allow=ulaw</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">dtmfmode=rfc2833</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">nat=no</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">fromuser=11234567890</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">extensions.conf</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[globals]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;…irrelevant stuff</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">trunk_1 = Dahdi/g1</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">trunk_2 = SIP/trunk_2</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">OUT_2 = SIP/bandwidth.com_outbound</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;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><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[frombandwidth]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;exten = _+1.,1,Set(GROUP()=SIPGROUP)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _+1.,n,Set(DID=${EXTEN:2})</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _+1.,n,Goto(DID_trunk_2,${DID},1)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;This is where it breaks. I tried to make it so there can't be more than 2 calls on SIP channels at once.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;Since it counts the phone as a channel, and adds it to the group, I had to use 4.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[internalphones]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,1,Set(GROUP()=SIPGROUP)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} >= 4]?100) ;If the group has 2 or more calls, do not dial.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,100,Playback(all-circuits-busy-now)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,101,congestion()</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _1NXXNXXXXXX,102,busy()</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">;This is where incoming calls go to if I'm awake.</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">[DID_trunk_2_timeinterval_Awake]</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _NXXNXXXXXX,1,Set(GROUP()=SIPGROUP)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _NXXNXXXXXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _NXXNXXXXXX,n,Set(CALLERID(num)=1${CALLERID(num)})</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">exten = _NXXNXXXXXX,n,Goto(voicemenu-custom-1|s|1)</span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> </span><o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">Thanks.</span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">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)<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<div>
<p class="MsoNormal">_______________________________________________<br>
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</blockquote>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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