On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <[EMAIL PROTECTED]>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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