Rings through Asterisk when dialing plain phones

Asterisk Asterisk Support
1

Hello,

I have an asterisk 1.4 setup with a TDM402B PCI in a dual xeon with 2gb ram. One FXO port on the card is plugged into my home POTS line. The POTS line also has several ‘dumb’ phones on it, or just plain old telephones. I also have a couple GXP-2000’s connected to a gigabit ethernet switch. My goal is to have it so that when a phone call comes in to my home line that the regular phones will ring and operate as normal and also that asterisk will not pick it up but rather will route the calls to the GXP-2000’s. This is working as planned.

However, when attempting to dial out on any of the plain phones something is causing asterisk to ring the GXP-2000 extensions as if a phone call was incoming. I can dial out using the GXP-2000’s just fine but if I attempt to use any regular phones asterisk gets wonky.

Has anyone run into something like this?

Here’s some debug log:
[Jun 20 16:39:52] DEBUG[4583] chan_zap.c: Requested indication 3 on channel Zap/3-1
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘s’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘fxo3-incoming’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘Zap/3-1’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘SIP/gxp20002-081c6728’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘Dial’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘SIP/gxp20001&SIP/gxp20002|20’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘2008-06-20 16:39:52’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘2008-06-20 16:40:00’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘8’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘0’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘FAILED’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘DOCUMENTATION’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘1213997984.14’
[Jun 20 16:40:00] DEBUG[4583] pbx.c: Function result is ‘’
[Jun 20 16:40:00] DEBUG[4583] chan_zap.c: Hangup: channel: 3 index = 0, normal = 20, callwait = -1, thirdcall = -
1
[Jun 20 16:40:00] DEBUG[4583] chan_zap.c: disabled echo cancellation on channel 3
[Jun 20 16:40:00] DEBUG[4583] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/3-1
[Jun 20 16:40:00] DEBUG[4583] chan_zap.c: Updated conferencing on 3, with 0 conference users

Here’s the relevant bit from zapata.conf:
rxgain=0.0
txgain=0.0
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
usecallerid=yes
usedistinctiveringdetection=no
context=fxo3-incoming
;;; use fxs signalling on the fxo ports
signalling=fxs_ks
echocancelwhenbridged=yes
echocancel=yes
echotraining=yes
echotraining=800
callerid=asreceived
sendcalleridafter=2
channel => 3

2

Upon further digging I see that the debug log message “Requested indication 3” apparently means that chan_zap believes it hears ringing on the line so it gets to function zt_indicate which then rings the SIP phones.

I’m at somewhat of a loss here because I believe the plain phone & asterisk connected to the same line should work for outbound dials from the plain phone. It also seems like I can’t be the only one that has encountered the problems either! At least I hope not.

I’m trying to avoid adding additional debug code and recompiling asterisk if at all possible so if anyone has a configuration with plain phones (non-SIP) and asterisk connected to POTS and has run into any weirdness when the plain phones go off-hook, please let me know!