tdm400p

Greetings all,

I’ve got a digium card with 4 FXO channels. I am unable to set any of the channels to perform outgoing calls. Inbound is working very well. I have added the trunk on the AMP, and have set up up an outbound route to make use of.
zapata.conf

[trunkgroups]

[channels]
language=en
echocancel=yes
echocancelwhenbridged=no
faxdetect=both
busydetect=yes
busycount=3

; Including file containing the suggested configuration
; generated by the hardware detector tool
; Remove this line if you don’t want this feature
#include zapata-channels.conf;
#include zapata_additional.conf;

zapata-channels.conf

; Autogenerated by /usr/sbin/genzaptelconf – do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;

; Span 1: WCTDM/0 “Wildcard TDM410P Board 1” (MASTER)
;;; line="1 WCTDM/0/0"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
;context=default

;;; line="2 WCTDM/0/1 RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 2
context=default

;;; line="3 WCTDM/0/2 RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
context=default

;;; line="4 WCTDM/0/3 RED"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
context=default

Your help is highly appreciated

to from-zaptel on all trunks.

cosmicwombat,

thanks for your suggestion, unfortunately it didnt work :frowning: … as far as i know, from-zaptel is dedicated only to receiving inbound calls from the pstn
which is already working well for channel 2. the problem is with outbounds

Hi,

I own a TDM400 too, with one [email protected] and one [email protected], my zaptel.conf is:


fxoks=1
fxsks=4
loadzone = us
defaultzone = us

After that, my zapata.conf:

signalling=fxs_ks
context=from-zaptel
language=fr
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
;echotraining=800
faxdetect=both
callprogres=yes
group=0
channel=>4

of course only for channel 4.

lokk at the context, as it’is from-zaptel

zydoon,

I’ve tried your settings too, didnt work either.

its frustrating

malazzeh,

You are confusing issues here.

The context is used for incoming calls. So that when a call comes in that is the context that it is assigned when it is entered into the system. So make all 4 channels the same context or things will start to work strangely as you can’t have different context’s in the same group.

Once you have done that you’ll need to restart asterisk so that it reads the updates and processes things correctly.

Now the reason you can’t make outbound calls is a different matter. That has to do with your outbound route. But you’ve provided no information on your outbound route. How it’s configured, dialing pattern, etc.

fskrotzki,

I have put a dial plan in the outbound routes with prefix 9|. to pick the digium pstn trunk for outgoing. I tried to assign any of the available channels in the trunk definition in order to make the outgoings. So what iam doing is assigning them by channels , not groups… but also to test, i have tried to assign 4 different groups, one for each channel. and tried to utilize a group no. in the trunk definition on the gui, but with the same negative result. nothing goes thru, on the cdr i dont see there is a hit on the zap.

Hi again,

I continue with my config:

I have a ZAP trunk called g0.
I assume that g0 is the set of FXOs included in group=0
than, an outbound route with the dial pattern “9|.” and pointing to the ZAp/g0 trunk

it’s working on my TDM400 and a TDM2422 of my customer.

I hope you find what missing,
Zydoon.

you can’t assign by channel, just by group. It will use attempt to use the last available in group first then work backwards (so in your case 4, then 3, then 2, then 1).

Dear all,

I have setup a new IP-pbx and found what the problem was. the problem happens because i have added the codec G729 to be used in sip.conf, since my extensions are all sip phones. and my VOIP provider supports only G729 for voice.
when i remove the allow=g729, the outgoing pstns work perfectly, otherwise the calls get the busy tone, and dont even hit the zap trunks.

Now the question is, How can i make th pstn accept G729 since its more efficient, has more features, and is supported by the voip provider.
If that cant be done, can i assign 2 different sip.conf files to be utilized when using either voip or pstn

Thnx

Actually, 729 places a higher CPU load in transcoding too. Unless you have a crowded network or remote phones on a poor connection, I don't see any compelling reasons to use 729. Anywhere.

 

btw, which provider only uses 729?

Yes, Remote connections are to be used, on mobile phones, that sometimes have to connect through 3G network, thus they are required to use a fixed amount of data according to their connection plan.

You mean, there is no way to translate the G729 back to sth else for PSTN, or another way to work it out

Haven’t tried this so don’t know if it works with ZAP.

edit zapata_additional.conf and put in

disallow=all
allow=g711

Save and reload asterisk

It certainly works for SIP trunks but some expert here will be able to help better probably.

Brian

In fact, while I don’t recall what PTSN uses, but I do not Asterisk has to transcode from G711, G729 or whatever to PTSN.

So, I am pretty sure therefor no G729 PTSN.

Correct. It is G711 aka ulaw/alaw depending on where you are located.

G729 requires licensing to be installed to work on anything but strick pass thru (aka, accept in on a sip trunk and pass to a sip phone).

I have trixbox with a card A200 with 4 ports FXO

the inbound routes works very good in all the ports, i tested with one analogical line one at a time on each port

i did the outbound routes one for local calls and other to allow long distance calls
when i plug the line on the port 1 the outbound calls works perfect
but whe i plug the line on the ports 2,3 and 4 the calls don`t go outside i just hear silence

in my gui i have the same like you ZAP/g0
the zapata file said is the group for the 4 channels but just one port works well for outside calls

I dont` know why or what to do, i am not to expert but i can do the instruction of any who want to help me, i am praying for help jejeje

regards!

I have trixbox with a card A200 with 4 ports FXO
the inbound routes works very good in all the ports, i tested with one analogical line one at a time on each port
i did the outbound routes one for local calls and other to allow long distance calls
when i plug the line on the port 1 the outbound calls works perfect
but whe i plug the line on the ports 2,3 and 4 the calls dont go outside i just hear silence in my gui i have the same like you ZAP/g0 the zapata file said is the group for the 4 channels but just one port works well for outside calls I dont know why or what to do, i am not to expert but i can do the instruction of any who want to help me, i am praying for help jejeje
regards!

Felon,
When you unplug port 1, then plug in to port 2, what do you do with port one? Could it be that your call from the PBX is trying to use port 1, but it’s not plugged in?
Is it possible to plug in 2 lines, then make a call that goes out your first port (to tie it up) then make another call? It should then go out port 2.

TomSyr

I am having this same problem with A400P and 3FXO modules.
When I make a call on port1 it goes throuogh fine. If I then make a call on port2 while port1 is active i cant get an outside line. Same for port3 until I habg up port1.

i plug the four lines and when trix box see busy the first port use the next one
so the free pbx already do all the config for me in one group named ZAP/g0
and is working prfect , my outbound routes and the incoming routes work great

now my consern is the next because we want to take all the traffic of celulars just by one trunk
and if the group have it all

how can just take one line for this purpose?