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
[EMAIL PROTECTED]: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
[EMAIL PROTECTED]: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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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, <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] at openSubscriber.com
>> http://www.opensubscriber.com/message/[email protected]/10416933.html
>>
>
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