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

[Kurt Knudsen]

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
>