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

Fri Oct 10 20:47:00 CDT 2008

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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>>>
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
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>>
>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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