[asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

[Steve Totaro]

On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <kurt.knudsen at gmail.com>wrote:

> Hello,
>
>
>
> 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:
>
>
>
> Users.conf (added by Asterisk-GUI)
>
> [trunk_2]
>
> provider = Bandwidth (SIP)  ; GUI metadata
>
> context = DID_trunk_2
>
> hasexten = no
>
> hasiax = no
>
> hassip = yes
>
> host = 216.82.224.202
>
> registeriax = no
>
> registersip = no
>
> usecallerid = yes
>
> nat = no ;Testing
>
> trunkname = Bandwidth.com (Sip)  ; GUI metadata
>
> username =
>
> secret =
>
> disallow = all
>
> allow = ulaw,alaw,g726
>
>
>
> sip.conf
>
> [general]
>
> context = frombandwidth
>
> ;other variables, etc.
>
>
>
> ;Added according to Bandwidth.com's wiki entry. Changed to inband because
> we were having DTMF issues.
>
> [bandwidth.com_inbound]
>
> host=216.82.224.202
>
> port=5060
>
> type=peer
>
> disallow=all
>
> allow=ulaw
>
> dtmfmode=inband
>
> canreinvite=no
>
> reinvite=no
>
> context=frombandwidth
>
> nat=no
>
>
>
> [bandwidth.com_outbound]
>
> host=216.82.224.202
>
> port=5060
>
> type=peer
>
> disallow=all
>
> allow=ulaw
>
> dtmfmode=rfc2833
>
> nat=no
>
> fromuser=11234567890
>
>
>
> extensions.conf
>
> [globals]
>
> ;…irrelevant stuff
>
> trunk_1 = Dahdi/g1
>
> trunk_2 = SIP/trunk_2
>
> OUT_2 = SIP/bandwidth.com_outbound
>
>
>
> ;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.
>
> [frombandwidth]
>
> ;exten = _+1.,1,Set(GROUP()=SIPGROUP)
>
> exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
>
> exten = _+1.,n,Set(DID=${EXTEN:2})
>
> exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
>
> exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
>
>
>
> ;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
>
> ;This is where it breaks. I tried to make it so there can't be more than 2
> calls on SIP channels at once.
>
> ;Since it counts the phone as a channel, and adds it to the group, I had to
> use 4.
>
> [internalphones]
>
> exten = _1NXXNXXXXXX,1,Set(GROUP()=SIPGROUP)
>
> exten = _1NXXNXXXXXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} >= 4]?100)  ;If
> the group has 2 or more calls, do not dial.
>
> exten = _1NXXNXXXXXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
>
> exten =
> _1NXXNXXXXXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
>
> exten = _1NXXNXXXXXX,100,Playback(all-circuits-busy-now)
>
> exten = _1NXXNXXXXXX,101,congestion()
>
> exten = _1NXXNXXXXXX,102,busy()
>
>
>
> ;This is where incoming calls go to if I'm awake.
>
> [DID_trunk_2_timeinterval_Awake]
>
> exten = _NXXNXXXXXX,1,Set(GROUP()=SIPGROUP)
>
> exten = _NXXNXXXXXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
>
> exten = _NXXNXXXXXX,n,Set(CALLERID(num)=1${CALLERID(num)})
>
> exten = _NXXNXXXXXX,n,Goto(voicemenu-custom-1|s|1)
>
>
>
> Thanks.
>   <http://lists.digium.com/mailman/listinfo/asterisk-users>


Is your Asterisk box on a public IP or behind a NAT/Firewall?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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