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

[Steve Totaro]

You need to configure your box for nat settings, externip and other settings
in sip.conf and set nat=yes instead of nat=no.

One way audio is almost always a NAT issue and those are two glaring things
that would cause problems.

Thanks,
Steve Totaro

On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen <kurt.knudsen at gmail.com>wrote:

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