Cannot make ZAP outbound calls

Eggars

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#1
Get "Cannot complete the call as dialed", when making call from ZAP phone.

This is my first Asterisk box with Elastix. I have been able to configure inbound calls seemingly correct (they work).

I am using a Rhino Channel bank with a crossover cable to a Zaptel Card in my aterisk box, and out to my T1.

I know you are going to need much more info so just point me in the right direction and I will get it ASAP.
 

dicko

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#2
How about you start with your outbound trunks? (SIP, ZAP) can you call station to station?
add the outbound routes you configured, and let's see . . .
 

Eggars

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#3
Zap Trunk g0
Outbound Caller ID "MyNumber"
Never Override CallerID "Not Checked"
Maximum Channels "Blank"
Disable Trunk "Blank"
Monitor Trunk Failures "Blank"
Dial Rules:
XX
911
1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
1NXXXXXXXXX

Outbound Prefix: "blank"
Zap Identifier "g0"

Yes, calls between extensions work zap to zap. i have no iax or sip set up.
 

Eggars

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#4
Outbound Route 0
Route Name Outbound
Route Password: "blank"
PIN Set:"none"
Intra Company Route:"blank"
Dial Patterns:"same as trunk"
Trunk sequence: 0 "ZAP/g0"
 

dicko

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#5
They look fine,

just to confirm
the station are dialing using
1NXXNXXXXXX?
no internal extension is numbered 1x.?

a question
XX in outbound route, why?

a comment,
down-line you should use the zap trunk with G0 not g0 to reduce what is known as "glare", this makes the calls go out from channel 24 downwards, (assuming the incomings come in upwrads of course), but only of course when you can dial out ;)

please post

/etc/zaptel.conf
and /etc/asterisk/zapata.conf
 

Eggars

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#6
No internal numbers beginning in 1

XX somehow was populated by default and I just have not removed it.

[trunkgroups]

[channels]
group=1
context=from-zaptel
signalling=em_w
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=4.0
txgain=2.0
callgroup=1
pickupgroup=1
immediate=no
switchtype=national
channel=>1-24
busydetect=yes
busycount=10

group=2
signalling=fxo_ks
rxwink=300
channel=>25-48

;Uncomment these lines if you have problems with the disconection of your analog lines
;busydetect=yes
;busycount=10



#include zapata_additional.conf
#include zapata-channels.conf
 

Eggars

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#7
etc/zaptel.conf

# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Global data

loadzone = us
defaultzone = us

span=1,1,2,esf,b8zs
span=2,0,0,esf,b8zs
e&m=1-24
fxoks=25-48
 

dicko

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#8
Apparently you have no group 0 and you try and dial out using this trunk

switchtype=national

is for PRI trunks and inappropriate here

busydetect=yes
busycount=10

is being applied to your extensions but not to your trunks where it might or might not be needed. (depending on the telcoo's disconnect supervision)
 

Eggars

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#9
without damaging my inbound route what would be best practice to correct this over sight?

by the way thank you for helping me understand how this all ties together!!
 

dicko

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#10
change g0 to G1 in outbound route
(you need to create a new trunk G1)
 

Eggars

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#11
can I not change the identifier of the previously id g0 to G1?
 

dicko

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#12
You could try ;)

trunk ZAP/g0 refers to trunk group 0 using the lowest channel first
trunk ZAP/G1 would refer to trunk group 1 using highest channel first
the second trunk would reflect your current settings more appropriately

context=from-zaptel
indicates a solution you applied which is probably unnecessary for your trunk, but if if works it doesn't need fixing. (I assume a block of numbers from the telco delivered with DID info, maybe I'm wrong)

threewaycalling=yes
transfer=yes

only appropriate if your telco supports "hook flash" service

rxgain=4.0
txgain=2.0
if possible check your csu/dsu settings, (competent telcos normally check the "normalization" before handing it off to you)
and

span=1,1,2,esf,b8zs

might need editing to match the line build out
 

Eggars

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#13
rx and tx gain I guess I was under the impression that this was internal distance settings. Is this not correct?

forgive me but I am not quite clear on what this last part means, I understand the B8ZS but the rest not so much. I really feel like a noob now.

Okay so I added the Trunk with the Identifier G1, and chose it as my first trunk in my outbound route. When I attempt to make an outbound call I get only silence the destination does not ring either. when I look at the Admin console of the unembbedded freePBX it shows me that 1 external call is active.

I re-auto configured the T1 settings on my Rhino just in case and there was no change. I am guessing I still need to edit the span...
 

dicko

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#14
rx and tx gain are the gains (in decibels) applied to the receive and transmit part of the calls, I have posted previously with specifics on how I use "zapmonitor" and "milliwatt" lines to normalize the gains.

The asterisk CLI will give you more detailed info as to call progress, the trunk behavior depends on how the trunks are provisioned by the telco, if they send you dialtone on wink then I believe your settings are correct and further investigation is required, if no dialtone is sent, try changing immediate to "yes" (sorry edited due to original typo) in zapata.conf. Is this a DID trunk?

editing zapata,conf rquires a complete asterisk restart for the effect to be applied
changes to zaptel.conf take a zaptel driver reload.
 

Eggars

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#15
DID trunk? Good question... If by the trunk in which calls are routed to from the toll frees I have asigned; yes. I only have one T1 connected to this setup.

do you know of a good guide on how to use the CLI? i have tinkered around a little with it but that is about it.
 

dicko

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#16
from your configs
signalling=em_w
this means ear&mouth with wink, if that's the signalling from the telco then it is appropriate, (other schemes are used less commonly)

DID:
"Direct Inward Dial" you buy maybe 20 numbers from the telco, the send over the trunk a wink, you answer, they send you the last few digits, you read these and route appropriately. (you would need immediate=no on this trunk for this to work)
No DID, is usually just one number from telco and they just send you a wink on any inbound call, you are expected to answer with a wink, the call is then connected (into the context you specified) . It will always be from that number. Only your Telco can confirm the provisioning on this trunk, you can try different things until it works however.
Outbound calling:

You wink the trunk, they wink back and if immediate=no then you wait for dialtone from the telco and proceed with spilling the DTMF's. If on winking the trunk there is no dialtone returned, then set immediate=yes and just send DTMF's after the returned wink.
CLI:
http://www.voip-info.org is my preferred reference.
 

Eggars

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#17
Sorry for the delayed response, been a busy day. I tried tinkering with these settings with no such luck. You mentioned that i can dig deeper with the cli... what do i need to do for it to monitor / trace and outbound call?
 

dicko

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#18
briefly:

set debug 99
and
set verbose 99

(don't even try to follow oon the CLI, instead follow, browse grep, whatever the log file "/var/log/asterisk/full" after setting the "noise level" to way loud )

get a detailed and authoritative specification of the trunk signalling and provisioning from the telco (don't take no for an answer, you pay for it, they are all bastards and will treat you like an idiot)

read the documentation on zapata.conf on voip-info.org
match all parameters to what the telco signalling is.

tools available:

zttool for following signalling on the channels on a span (yours is the first one)
ztmonitor for monitoring audio energy on a particular trunk

remember to restart asterisk after ANY edits to zapata.conf
add a reload of zaptel if zaptel.conf is edited.

Don't give up till you make it work (inbound works so at least you have a start)

keep me posted.
 

Eggars

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#19
i noticed in the unembedded freePBX that it says i have 3 bad destinations. but how can that be... if inbound is working... very strange
 

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