Tried using GROUP()?
When a call comes in or goes out:
Exten => XXX,1,Set(GROUP(bdwi_out_1)=outgoing/incoming);
Exten => XXX,n,GotoIf($[${GROUP_COUNT(outgoing/[EMAIL PROTECTED])}] > 1?fail)
Exten => XXX,n,Dial(...)
Exten => XXX(fail),1,Congestion();
Exten => XXX(fail),n,Hangup();
Obviously choose outgoing or incoming, if you want to track both you can just
use $MATH() to add them together.
Or some other math logic to check the result.
On incoming Set(DIALSTATUS=CHANUNAVAIL) and it'll ring busy to bandwidth(or out
of service, you can tweak this).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
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]<mailto:[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]<mailto:[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]<mailto:[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]<mailto:[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]<mailto:[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]<mailto:[EMAIL
PROTECTED]>> wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <[EMAIL PROTECTED]<mailto:[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<http://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<http://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<http://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.
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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