Thanks, Steve, That's what I am unsure of. I don't know how to limit 1 call per trunk. If that's an easy thing to setup, I'd love to see it.
On Fri, Oct 10, 2008 at 10:20 PM, Steve Totaro < [EMAIL PROTECTED]> wrote: > Oh, I thought you had logic to count the calls on the trunk. You should > limit each trunk to one call. This is the primary reason besides the email > that basically said that customer support structure has been changed and > anything beyond the Demarc would not be supported, I the Demarc is simply > their boxen, so unless it is on their side, you will not get any helpful > support from Bandwidth, plus they CCed over 500 people by address instead of > setting up a group. > http://www.bandwidth.com/content/support/?page=standardSupport > > I am with Junction and while a trunk is not "unlimited" as far as price for > usage, the amount of trunks is unlimited (or at least as unlimited as it can > be since nothing is really unlimited). They asked that I try not to go over > one call per second for any real duration, and that I not hammer one LATA do > to limited interconnects. > > The other thing was Junctions was very easy to sign up with, great support, > and configuration was a breeze. > > As for Bandwidth, I think they are solid but due to recent changes and the > fact that you must pay per channel, as well as the setup process, I decided > they were not for me. > > I will take a second look at your sip.conf and extensions.conf later to see > if something jumps out at me. I suspect since you are setting up two > separate trunks with Bandwidth, you need to limit each trunk to one call, > rather than two. > > Thanks, > Steve Totaro > > > > > On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen <[EMAIL PROTECTED]>wrote: > >> 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 < >> [EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]>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 < >>>> [EMAIL PROTECTED]> wrote: >>>> >>>>> >>>>> >>>>> 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) >>>>> >>>>> >>> > > _______________________________________________ > -- 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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