Any updates? It still seems to happen, though not as often as it used to. We're using Polycom 320 phones, if that makes a difference, though we did do it with X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen <[EMAIL PROTECTED]>wrote: > 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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