One of my 7941's arrived today, meaning I could use a 'real' handset now in my testing in relation to this problem.

What I discovered is that I am not getting any adio for an incoming call from the PSTN as well (ie PSTN->VoIP) on top of the previous problem of not being able to dial out (maybe this helps isolate the problem)

I'm quite confused with PLAR and how it is meant to work.

At this stage, all I'm trying to do is:

1) direct calls from the PSTN to the 7941 (extension 1002)

2) be able to dial out from the 7941 to a PSTN number

Here is my running-config

!
voice-port 3/0
supervisory disconnect dualtone mid-call
no comfort-noise
cptone AU
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing digit 80
timing inter-digit 80
timing min-ring 200
connection plar 1002
impedance complex1
description Connected to the PSTN
!
voice-port 3/1
impedance complex1
!
!
!
!
!
!
dial-peer voice 1 pots
description 10 Digit Mobile Dialling
destination-pattern 04........
no digit-strip
port 3/0
!

!
telephony-service
load 7902 CP7902080002SCCP060817A
load 7941GE SCCP41.8-3-5S
load 7941 SCCP41.8-3-5S
load 7961GE SCCP41.8-3-5S
load 7961 SCCP41.8-3-5S
max-ephones 8
max-dn 16
ip source-address 10.1.1.1 port 2000
time-format 24
date-format dd-mm-yy
max-conferences 4 gain -6
transfer-system full-consult
login timeout 60
create cnf-files version-stamp 7960 Oct 02 2009 21:35:45
!

!
ephone-dn  3
number 1002
!

!
ephone  3
description Cisco IP Phone 7941
mac-address 0018.73C0.C2E5
type 7941
button  1:3
!

With this configuration when I dial out (to a number beginning with 0422...)

-the 7941 generates 1 ringback sequence and then is in a 'empty' call.  The called PSTN extension never rings.

When I make a call from my mobile to the PSTN line attached to the CME (there is also a normal analogue phone in addition to the CME attached to the line)

-both the 7941 and the analogue phone ring for 1 sequence

-The analogue phone stops ringing, but the 7941 continues to ring (I believe this is because the CME itself answers the call after 1 ring)

-At this stage the calling mobile is in an 'empty' call with the CME

-If I pick up the ringing 7941, it is also is in an empty call (and cannot hear the mobile)

Both the CME and 7941 are in the same subnet/VLAN.

I obviously have something seriously wrong with the configuration

Hi Colin

Thanks for your response and suggestions.

I will stick to trying to get the PLAR configuration working as it was something I thought would be useful for CCNA Voice (640.460) exam that I am taking next week.  What has become increasingly obvious is that the 1 page explanation (p 271) and suggested configuration that the Cisco Press book (Cioara) dedicates to FXO PLAR Configuration is not sufficient.

My understand of PLAR is as follows:

-ports configured with PLAR automatically dial a designated number as soon as an off-hook signal is detected

-this allows an emergency/bat style phone to be implemented wherebouts just lifting the handset will dial the designated number

-the reason PLAR needs to be used with respect to FXO is because when a call comes in on the PSTN to the CME, the CME needs to know which handset to deliver the call to (as it can potentially have multiple handset with multiple extension numbers)

-I believe the handset could be an analogue/VoIP unit and it is probably possible to assign more than one handset to the designated PLAR extension (to mimic the behaviour where you have more than one POTS phone connected to your PSTN line resulting in all the handsets ringing); a shared line experience

So, I believe we are just using PLAR to handle the incoming call from the PSTN and can dial out via:

(i) picking up a VoIP handset (possibly using a different line) and dial a PSTN number which will be dialed via DTMF/pulse

(ii) dialling an access number '9' (possibly using a different line) to get the PSTN dialtone (instead of CME dialtone) and then dial via DTMF/pulse

I altered my configuration as per your description, other than maintaing the line characteristics for Australian PSTN (below) and my existing 10 digit mobile pots dial-peer

!
voice-port 3/0
cptone AU
connection plar opx 1003
impedance complex1
description Connected to the PSTN
!
voice-port 3/1
impedance complex1
!
!

!
dial-peer voice 82 pots
destination-pattern 82

port 3/0

!

!
dial-peer voice 1 pots
description 10 Digit Mobile Dialling
destination-pattern 04........
no digit-strip
port 3/0
!

!
ephone-dn  3
number 1002
name Manager
!
!
ephone-dn  4  dual-line
number 1003
name Private Line
trunk 82
!

!
ephone  3
mac-address 0018.73C0.C2E5
button  1:3 2:4
!

Based on my incomplete understanding of your configuration I believe:

-incoming calls from the PSTN would come in on line 2 (1003)

-I would make outgoing calls to the PSTN using line 1(1002)

Result:

1) I called the PSTN from my mobile, Line 2 rang (1003) when I answered it, I had the 'empty' (CME bridged) no audio call

2) Pressing the Line 1 button (1002), and dialling a my mobile extesion (04........) I still get into the 'empty' bridge call that CME does before (see below)

*Apr 21 09:02:30.066: htsp_process_event: [50/0/3.1, EFXS_ONHOOK, E_DSP_SIG_1100
]efxs_onhook_offhook htsp_setup_ind
*Apr 21 09:02:30.066: [50/0/3.1] get_local_station_id calling num=1002 calling n
ame=Manager calling time=04/21 17:02  orig called=
*Apr 21 09:02:30.086: htsp_process_event: [50/0/3.1, EFXS_WAIT_SETUP_ACK, E_HTSP
_SETUP_ACK]efxs_check_auto_call
*Apr 21 09:02:31.043: htsp_digit_ready(50/0/3.1): digit = 0
*Apr 21 09:02:31.484: htsp_digit_ready(50/0/3.1): digit = 4
*Apr 21 09:02:31.941: htsp_digit_ready(50/0/3.1): digit = 5
*Apr 21 09:02:32.273: htsp_digit_ready(50/0/3.1): digit = 6
*Apr 21 09:02:32.814: htsp_digit_ready(50/0/3.1): digit = 2
*Apr 21 09:02:33.147: htsp_digit_ready(50/0/3.1): digit = 3
*Apr 21 09:02:33.515: htsp_digit_ready(50/0/3.1): digit = 1
*Apr 21 09:02:34.108: htsp_digit_ready(50/0/3.1): digit = 4
*Apr 21 09:02:34.465: htsp_digit_ready(50/0/3.1): digit = 5
*Apr 21 09:02:34.757: htsp_digit_ready(50/0/3.1): digit = 6
*Apr 21 09:02:34.773: htsp_timer_stop3
*Apr 21 09:02:34.777: htsp_process_event: [50/0/3.1, EFXS_OFFHOOK, E_HTSP_PROCEE
DING]efxs_offhook_proceeding
*Apr 21 09:02:34.777: [50/0/3.1] set signal state = 0x8 timestamp = 0htsp_setup_
req
*Apr 21 09:02:34.781: htsp_process_event: [3/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]f
xols_onhook_setup
*Apr 21 09:02:34.781: [3/0] set signal state = 0xC timestamp = 0
*Apr 21 09:02:34.785: htsp_timer - 1300 msec
*Apr 21 09:02:36.087: htsp_process_event: [3/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVE
NT_TIMER]fxols_wait_dial_timer  htsp_dial
*Apr 21 09:02:38.095: htsp_process_event: [3/0, FXOLS_WAIT_DIAL_DONE, E_DSP_DIAL
ING_DONE]fxols_wait_dial_done htsp_progress
*Apr 21 09:02:38.099: htsp_timer - 350 msec
*Apr 21 09:02:38.099: htsp_call_bridged invoked

2) Pressing the Line 3 button (1004), attempts to send '82' out the PSTN line.

So I think the 'trunk' command is used to tie a specified number to particular line (ie PLAR on the outward leg, which is not something I want to do).  I want to be able to use the VoIP service so I can get more familiar with it (this necessitates being able to make outbound calls to any PSTN number like a traditional analogue phone)

Does this mean I have to use STCAPP as you suggested and the PLAR stuff is completeley foo bar?

Kind Regards

Ben