[asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

[Jeremy Mann]

My experience with GotoIf, what follows the ? has to be part of the extension itself.

In your example:

Exten = _1NXXNXXXXXX(100) would be the intended target.

Maybe that's just 1.4 specific, I'll admit I haven't read this entire thread.

Also, use specific groups:

Set(GROUP(SIP)=SIPGROUP)

Set(GROUP(SIP_PHONE)=SIPGROUP)

Those are two distinct ways to track them, instead of a general GROUP() statement.  Since a channel can only be a member of one GROUP(), but multiple GROUP(XXX) it makes it easier to track items when they belong to multiple things(and logically reads better for future support of the dialplan).

I can say that we're successfully limiting calls on two-way sip trunks from bandwidth, both incoming and outgoing.

Probably if 4+ lit up at once I'd have a problem, but we're not that high volume.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio

I tried using GROUP(), here's a snippet from the first post.

;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)

I'll try playing around with incoming/outgoing and see if that makes a
difference. I don't know why it counts the phone as a channel, though.

On Mon, Oct 20, 2008 at 12:14 PM, Jeremy Mann <jmann at txhmg.com> wrote:
>
> 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/incoming at bdwi_out_1)}] > 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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 <kurt.knudsen at gmail.com> 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 <stotaro at totarotechnologies.com> 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 <kurt.knudsen at gmail.com> 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 <stotaro at totarotechnologies.com> 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 <kurt.knudsen at gmail.com> 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 <stotaro at totarotechnologies.com> wrote:
>
>
>
> On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen <kurt.knudsen at gmail.com> 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.
>
>
>
> 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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