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 >>> >> >> >> _______________________________________________ >> -- 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 >> > > > > -- > 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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