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)
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