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

Mon Oct 20 14:48:05 CDT 2008

Well, when it fails over to the Dahdi trunk, it doesn't dial properly,
so I think I broke the macro. I will add the Set(GROUP()) stuff inside
of that macro-trunkdial-0.3 context and see if that helps. But it's
weird that I can't dial out. Here's a bit of the full log:

DEBUG[8221] app_macro.c: Executed application: Dial
VERBOSE[8221] logger.c:     -- Executing
[1-dial at macro-trunkdial-failover-0.3:2] GotoIf("SIP/207-0a1b3590", "20
> 0 1-CONGESTION|1:1-out|1") in new stack
VERBOSE[8221] logger.c:     -- Goto
(macro-trunkdial-failover-0.3,1-CONGESTION,1)
DEBUG[8221] app_macro.c: Executed application: Gotoif
VERBOSE[8221] logger.c:     -- Executing
[1-CONGESTION at macro-trunkdial-failover-0.3:1] Dial("SIP/207-0a1b3590",
"Dahdi/g1/18005551212") in new stack
DEBUG[8221] dsp.c: dsp busy pattern set to 500,500
DEBUG[8221] chan_dahdi.c: Dialing '18005551212'
DEBUG[8221] chan_dahdi.c: Deferring dialing...
VERBOSE[8221] logger.c:     -- Called g1/18005551212
DEBUG[8221] chan_dahdi.c: Sent deferred digit string: T18005551212w
DEBUG[8221] chan_dahdi.c: Done dialing, but waiting for progress
detection before doing more...
VERBOSE[8221] logger.c:     -- Hungup 'DAHDI/1-1'

Not sure how it broke, but it won't use the Dahdi channel :( It just
goes to a busy signal after you dial. I tested on an analog phone and
it can dial out normally, so it's the system.

Thanks.

On Mon, Oct 20, 2008 at 2:29 PM, Jeremy Mann <jmann at txhmg.com> wrote:
> I have a macro to dial out, similar to yours in that it fails over to Zap/Dahdi trunks in the event our bandwidth stuff is overloaded.
>
> I run this in a macro, and only set and check groups within that macro.  I'm confused why yours would attach to "phones" in any way, unless you mean phone to phone calls, in that case don't set the group?
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kurt Knudsen
> Sent: Monday, October 20, 2008 1:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
>
> The GotoIf works, because it does failover sometimes, just not all the
> time, I followed instructions from here:
>
> http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
>
> And it seems to work in other areas that I use it in a similar way. I
> only have the Set(GROUP()) when we are making outgoing calls on the
> SIP trunk or when there's an incoming call on the SIP trunk. Anything
> on Dahdi doesn't get included. I don't know how to tell my phones and
> channels apart, I'm not trying to add the phones to the group, just
> the channels. Can you paste some of your extensions.conf since you
> also use Bandwidth.com?
>
> Thanks.
>
> On Mon, Oct 20, 2008 at 8:30 PM,  <zozo-007 at hotmail.com> wrote:
>> -- Kurt Knudsen 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.
>>
>> --
>> This message was sent on behalf of zozo-007 at hotmail.com at openSubscriber.com
>> http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10416933.html
>>
>
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