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

[Kurt Knudsen]

Thanks, Steve,

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.

On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

> 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.
> http://www.bandwidth.com/content/support/?page=standardSupport
>
> 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.
>
> The other thing was Junctions was very easy to sign up with, great support,
> and configuration was a breeze.
>
> 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.
>
> 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.
>
> Thanks,
> Steve Totaro
>
>
>
>
> On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen <kurt.knudsen at gmail.com>wrote:
>
>> 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.
>>
>> 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.
>>
>> Thanks :)
>>
>>
>> On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro <
>> stotaro at totarotechnologies.com> wrote:
>>
>>> 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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