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

Kurt Knudsen kurt.knudsen at gmail.com
Mon Oct 20 10:46:16 CDT 2008 

Any updates? It still seems to happen, though not as often as it used to.
We're using Polycom 320 phones, if that makes a difference, though we did do
it with X-Lite as well.

On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen <kurt.knudsen at gmail.com>wrote:

> 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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>
>
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