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

[Steve Totaro]

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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