NO JOY No inbound DAHDI progress ring - need help adding Playtones(ring) properly

Capstone Works

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So my problem is this:

For SIP Inbound calls in context [from-trunk-sip-nexVortex] , callers SUCCESSFULLY hear the ringing before the call is picked up by an extension or ring group

For DAHDI (Grande T1 PRI) Inbound calls in context [from-digital], callers hear SILENCE while the extension or ring group rings, until the call is picked up by an person or voicemail.

In sip_custom.conf we've tried various combinations of
pedantic=no
prematuremedia=no
progressinband=yes
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
ignoreregexpire=0

in sip_general_additional we've tried
<snip>
progressinband=yes
pedantic=no
<snip>


No luck with any of these sorts of settings - SO we moved on to DialPlan modification with SUCCESS, but not FreePBX tolerant success....

In summary, we were able to add exten => 8922,n,Playtones(ring) DIRECTLY to extensions_additional.conf, but we have not been successful in adding it to extensions_custom.conf in a way that would still let us MANAGE our extension definitions from within FreePBX

Any help would be appreciated!
Here goes the details:

------------------
Sanitized summary details, below.
<----------------------------------------------------->
Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE x
SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE x
Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE x
Disk Free = ADEQUATE| Mem Free = ADEQUATE | NTPD = ONLINE x
SendMail = ONLINE | Samba = ONLINE | Webmin = ONLINE x
Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A x

PIAF Installed Version = 2.0.6.5 under *HARDWARE* x
FreePBX Version = 2.11.0.38 x
Running Asterisk Version = 11.10.2 x
Asterisk Source Version = 11.8.1
Dahdi Source Version = 2.9.0
Libpri Source Version = 1.4.14
IP Address = x.x.x.x on eth0 x
Operating System = CentOS release 6.5 (Final) x
Kernel Version = 2.6.32-431.1.2.0.1.el6.x86_64 - 64 Bit x
Incredible Version = 11.8
CDR-MYSQL = OK | CDR-CUSTOM = OK | Flite Engine = OK x
G722 Codec = OK | G726 Codec = OK | Speex Codec = OK x
Resample Codec = OK | Alaw Codec = OK | Ulaw Codec = OK x
GSM Codec = OK | ILBC Codec = OK | Dahdi Codec = OK x
Jabber Connect = N/A | Gtalk Channel = N/A | Silk Codec = OK x
Siren7 Codec = OK | Siren14 Codec = OK | G729 Codec = N/A x
SCCP-B Codec = N/A | Motif Codec = OK | x

Asterisk Uptime = 10 minutes, 51 seconds x
System Uptime = 72 hours x
IAX2 Registrations = 0 IAX2 registrations. x
SIP Registrations = 0 SIP registrations. x
Parked Calls = 0 parked calls in total. x
Installed ISO Version = 20650 x
System Installed on = 2014-03-17T14:16-0500 x
Kickstart Method = Normal ks kickstart
<----------------------------------------------------->
In extensions.conf
; from-digital:
;
; Context to set for PRI's and equivalent
;
[from-digital]
exten => s,1,Playtones(ring)
include => from-digital-custom
include => from-pstn
;-------------------------------------------------------------------------------
<----------------------------------------------------->

In extensions_additional.conf
[from-trunk-sip-nexVortex]
include => from-trunk-sip-nexVortex-custom
exten => _.,1,Set(GROUP()=OUT_19)
exten => _.,n,Goto(from-trunk,${EXTEN},1)
;--== end of [from-trunk-sip-nexVortex] ==--;
[ext-did-0002]
include => ext-did-0002-custom <= Added this to try to add Playtopnes(ring) to this entire context (see below), but it does not work
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})
<snip>
exten => 8922,1,Set(__FROM_DID=${EXTEN})
exten => 8922,n,Gosub(app-blacklist-check,s,1())
exten => 8922,n,Set(CDR(did)=${FROM_DID})
exten => 8922,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 8922,n,Set(CHANNEL(musicclass)=default)
exten => 8922,n,Set(__MOHCLASS=default)
exten => 8922,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 8922,n,Set(CALLERPRES()=allowed_not_screened)
exten => 8922,n,Playtones(ring) <= Manually added this, it works, but FreePBX will of course overwrite
exten => 8922,n(dest-ext),Goto(from-did-direct,126,1)
<snip>
;--== end of [ext-did-0002] ==--;

[ext-did]
include => ext-did-custom
include => ext-did-0001
include => ext-did-0002
exten => foo,1,Noop(bar)
;--== end of [ext-did] ==--;
<----------------------------------------------------->
Calls come in via DHADI in context [from-digital]
[from-digital]
exten => s,1,Playtones(ring) <= Tried it here also, but this does not see to hav and effect
include => from-digital-custom
include => from-pstn
[from-pstn]
include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations
include => ext-did
include => ext-did-post-custom
include => from-did-direct
include => ext-did-catchall ; THIS MUST COME AFTER ext-did

<----------------------------------------------------->
In extensions_custom.conf, tried adding these

[ext-did-0002-custom]
exten => s,1,Playtones(ring)
;# // END ext-did-0002-custom
[ext-did-custom]
exten => s,1,Playtones(ring)
;# // END ext-did-custom


Thanks again for taking a look at this - its driving me crazy!
 

Capstone Works

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So I gave it a swing at doing it "right" reran dahdi_genconf, added the progress=yes in chan_dahdi_custom.conf and changed pridialdplan and prilocaldialplan to local, context from-digital (default settings) with no luck. Then on a whim, in system.conf I changed loadzone=us, defaultzone=us to loadzone=uk, defaultzone=uk, and now I have UK ring tones on inbound calls... question is, what is wrong with the US ;-)

(a can of worms to be sure, but the focus of my quesiton is DAHDI specific.)
 

rb1kenobi

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Hey bud, I have the EXACT same issue. If I use the UK ringtones, it works fine. Did you find a resolution (besides using the funky UK rinbgack tones? which I'm starting to warm up to.....)
 

Capstone Works

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Nope - still using UK, but the client want to move back to the US! :)

Please keep me posted - weird at best.
 

james

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Try adding
priindication = outofband

Also what version of Asterisk?
 

Capstone Works

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Thanks for the suggestion - try adding it to which conf file? chan_dahdi.conf?

Asterisk 11.10.2
 

Capstone Works

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and why would that only affect US indications? When set to UK it works fine.
 

james

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I dunno... I pokes #asterisk-dev on freenode to see if they had thoughts. I will look at the code tomorrow when I'm back at my desk.

Yes Chan_dahdi.conf
 

Capstone Works

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We did - several times up to and including an RMA on the card - we also reached out to the PRI vendor, with no luck - both hold that they support the function of the card or the circuit - the ringtones are created by the Asterisk settings and sent to the caller on the PRI channel - which we agree is what happens - just not when the US indications are selected.
 

rb1kenobi

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Just an fyi for all involved. I took this up with Digium support yesterday, and they said "it's Asterisk, not Digium"
See response below:

Created By: Adam Chandler (9/18/2014 3:55 AM)
-The card will unfortunately, not have anything to do with the ringback tone you hear.
-This is all handled by Asterisk so support for this will be under an L2 OSS or higher and is not covered under the Telephony card support period.
-You can look through the following google search and should be able to find some information that will help in getting this set up:

https://www.google.com/webhp?source...spv=2&ie=UTF-8#q=UK Ringback tone in asterisk
 

Capstone Works

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Same response I received from Digium, and I think they are right, as the UK indications DO work, US does not - same card same everyting else.
 

rb1kenobi

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Yeah, I'm pretty sure it's gonna be in the Chan_dahdi somewhere. I'm doing a lot of digging in some of the parameter files. I think I will try a few things over the weekend with the chan_dahdi.conf (and custom files to bypass freepbx).
I've got a theory after looking at this: http://doxygen.asterisk.org/trunk/chan_dahdi.conf.html
In particular
; Set the tonezone. Equivalent of the defaultzone settings in
; /etc/dahdi/system.conf. This sets the tone zone by number.
; Note that you'd still need to load tonezones (loadzone in
; /etc/dahdi/system.conf).
; The default is -1: not to set anything.
;tonezone = 0 ; 0 is US
 

tbrummell

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My PRI's have this.....
/etc/asterisk/chan_dahdi_groups.conf
Code:
; [span_1]
signalling=pri_net
switchtype=national
pridialplan=national
prilocaldialplan=national
group=0
context=from-digital
channel=>1-23
 
 
; [span_2]
signalling=pri_net
switchtype=national
pridialplan=national
prilocaldialplan=national
group=0
context=from-digital
channel=>25-47
group=1
context=from-digital
channel=>26-47
/etc/dahdi/system.conf:
Code:
span=1,1,0,ESF,B8ZS
span=2,1,0,ESF,B8ZS
bchan=1-23,25-47
dchan=24,48
fxsks=49-60
echocanceller=oslec,49-60
fxoks=61-72
echocanceller=oslec,61-72
loadzone=us
defaultzone=us
My callers hear ringback when calling in to the PRI.
 

rb1kenobi

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tbrummel: what card(s) are you using, and which module is modprobed for it?
 

tbrummell

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I did all of my config (the last time) with the FreePBX Dahdi Config module....
Capture.PNG
 

tbrummell

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Code:
root@pbx:~ $ dahdi_hardware
pci:0000:09:08.0    wct4xxp+    d161:1220 Wildcard TE220 (5th Gen)
pci:0000:10:09.0    wctdm24xxp+  d161:2400 Wildcard TDM2400P
 

rb1kenobi

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Hmm, yeah, unfortunately this may be an apples to potatoes comparison. I am using the TE133F

And in /etc/dahdi/system.conf
This works: (ie: call progress tones are heard, albeit with the UK sound)
span=1,1,0,ESF,B8ZS
bchan=1-23
dchan=24
loadzone=uk
defaultzone=uk


And this DOESNT work: (no call progress tones- complete silence)
span=1,1,0,ESF,B8ZS
bchan=1-23
dchan=24
loadzone=us
defaultzone=us

It can't be framing, it can't be signalling, it can't be dialplan etc etc otherwise it wouldn't work with every OTHER timezone besides US. right?
 

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tbrummell

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Hmmm, I do see a major difference. I'm configured as Network on the Digium/Asterisk side as I have AudioCodes Mediant 1000's hanging off the other end of the PRI converting to SIP....application specific for my purpose. That could be a huge difference between the two. I could try and reverse the setup (change Asterisk to CPE and Mediants to Network) given some time. Maybe tomorrow afternoon if things quiet up around here.
 

rb1kenobi

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Yes, I'm direct to telco (so I'm CPE). And for what it's worth, I did take it up with my telco, and they really don't have anything to do with ringback/call progress on the T1. They set framing, signalling, and # of channels, that's pretty much it. If I had that wrong, nothing would work, although plenty of people seem to want me to blame the telco.
Hey Capstone, if you're still following this thread, what model of Card are you using? (and which module does asterisk use on it?)
 
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