No outgoing Dahdi calls

Asterisk Asterisk Support
1

Hi,
It’s been a while since I altered my asterisk setup, but I’ve had a need to add a pstn connection. Previously the setup just used SIP/IAX trunks and that worked fine.
I’m using a X100P clone which I had lying around.
I’ve rebuilt asterisk/dahdi to include a rolling buffer for ukcallerid support and incoming is now working fine (including CID).
For the life of me I can’t get outgoing (pstn) calls to work.

Relevant conf files:

chan_dahdi.conf:
[trunkgroups]

[channels]
loadzone=uk
defaultzone=uk
usecallerid=yes
ukcallerid=yes
cidsignalling=v23 ; Added for UK CLI detection
;cidstart=polarity ; Added for UK CLI detection
cidstart=usehist
sendcalleridafter=1
callerid=asreceived ; propagate the CID received from BT.

context=from-pstn
signalling=fxs_ks
;channel=1
rxwink=300
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
;echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1

immediate=no

#include dahdi-channels.conf

dahdi-channels.conf:
;;; line="1 WCFXO/0/0"
signalling=fxs_ks
callerid=asreceived
group=1
context=from-pstn
channel => 1
callerid=
group=
context=default

relevant line from extensions.conf
exten => _90[1-9].,1,Dial(DAHDI/1/${EXTEN:1})

I’ve tried with DAHDI/g1 and DAHDI/1
Both produce the following in the cli:
– Executing [9@home:1] Dial(“SIP/2001-0000002d”, “DAHDI/1/”) in new stack
– Called 1/
– DAHDI/1-1 answered SIP/2001-0000002d

I don’t get why it is showing DAHDI/1-1 answered.
Any ideas where I’m going wrong ??

Rgds,
Jon

2

You haven’t specified answer supervision should be used. You are probably lucky that BT do reverse polarity on answer for domestic lines. Many PTTs provide no answer supervision at all on such lines, which is why the default is to assume answer.

3

Adding callprogress=yes & busydetect=yes to chan_dahdi.conf has changed the error.

In the cli I’m now seeing:
– Executing [@home:1] Dial(“SIP/2001-0000004d”, “DAHDI/1/”) in new stack
– Called 1/
– Hungup ‘DAHDI/1-1’
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/2001-0000004d’ status is ‘CHANUNAVAIL’

Now why on earth does asterisk think the line is busy ?

4

This has progressed somewhat - though I’m not aware I did anything.
outgoing calls are now connecting, but 9 times out of 10 there is no audio (in either direction).

5

seems that it didn’t like my txgain=15 setting. Set it to txgain=15.0 and all is now working fine.

6

The answer supervision option for BT lines is answeronpolarityswitch.