All outgoing PSTN calls dropped immediately

Asterisk Asterisk Support
1

Hi everybody,

I would appreciate your help. I have got lovely small server with asterisk on as well as OpenVOX TDM400P card with 4 FXO ports. All incoming calls are fine (except two-ring delay to show caller id), howver I am not able to make any outgoing call as they are dropped as soon as the other party answers the phone.

Here is output of the asterisk cli:

-- Zap/1-1 answered SIP/21-08f6b918

== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
– Hungup ‘Zap/1-1’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/21-08f6b918’ in macro ‘dialout-trunk’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/21-08f6b918’
– Executing [h@macro-dialout-trunk:1] Macro(“SIP/21-08f6b918”, “hangupcall|”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/21-08f6b918”, “w”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/21-08f6b918”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/21-08f6b918”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/21-08f6b918”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/21-08f6b918”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“SIP/21-08f6b918”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/21-08f6b918’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/21-08f6b918’
== Starting post polarity CID detection on channel 1
– Starting simple switch on ‘Zap/1-1’
– Hungup ‘Zap/1-1’

and here is my zapata:

;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity ; Added for UK CLI detection
answeronpolarityswitch=no
sendcalleridafter = 0
callerid=asreceived ; propagate the CID received from BT

language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

hanguponpolarityswitch=yes

;Include genzaptelconf configs
#include zapata-channels.conf

;Include AMP configs
#include zapata_additional.conf

Many thanks for your help.

Oliver

2

Hi

Comment out hanguponpolarityswitch=yes or set it to no it does sound like this may be the issue

Ian

3

Ian,

You are right, I have commented out hanguponpolarityswitch=yes and the calls are not dropping.

However, when the other party ends the call our side will not detect it and the call stays active.

The second problem is that without this switch asterisk will keep ringing about twice after the other party ends the call. For example, I do a test call to my office, hang up after 2 rings but asterisk will not detect it and keep ringing even when I have already hangup.

hanguponpolarityswitch=yes sorts this problem and asterisk will hangup immediatelly.

Any ideas?

Many thanks.

Oliver

4

Hi

Ok you need to get you multimeter or scope out.

It sounds like you lines are using polarty revesal. you can check this by putting the meter across the line and making a call. if it reverses on answer and then again on hangup or you get a small break in line current at teh end.

also try

busydetect=no answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=no

in your zapata.conf

Ian