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

[Kurt Knudsen]

Hi Steve,

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.

Thanks.

On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

>
>
> 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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