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14281 Views 16 Replies Latest reply: Feb 27, 2011 7:53 PM by Vikrant RSS 1 2 Previous Next

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FXO Problem

Oct 9, 2009 7:07 PM

Ben 25 posts since
Aug 2, 2009

Hi

 

I'm having a problem making an outbound call (VoIP -> PSTN) using a VIC-2FXO card in a 1760V (running 12.4(15)T7) and was hoping someone might be able to point me in the right direction.

 

-The 1760V is running CUCME 4.1 (10.1.1.1/24)

-I am using a Blue Ip Softphone to initiate the call

-I can successfully answer a call from the PSTN on the VoIP device

-the PSTN line is plugged into port 0 of the VIC-2FXO card (slot 3) via an RJ-11 cable

-if I plug a normal analogue handset into the same RJ-11 cable coming into the VIC-2FXO I can make a call

-I can use DTMF from an anlogue phone to dial out (hence have set dial type to DTMF)

 

Here are the relevent bits of running-config:

 

!
voice-port 3/0
supervisory disconnect dualtone mid-call
pre-dial-delay 2
ring number 2
no echo-cancel enable
cptone AU
timeouts call-disconnect 3
timeouts ringing infinity
timeouts wait-release 3
timing min-ring 200
connection plar 1001
impedance complex1
description Connected to the PSTN
!
voice-port 3/1
impedance complex1
!
!
!
!
!
!
dial-peer voice 1 pots
destination-pattern .T
no digit-strip
port 3/0
!

 

Via a #debug vpm signal, I believe the line is going OFF Hook and the VIC-2FXO/CME is sending the dialled digits via DTMF, however the PSTN number never rings and the CME appears to do some bridging fullback feature where the extension ends up in an 'empty' call with the CME acting as the called endpoint (10s timeout).  See attached debug trace.

 

I think the problem could be either:

-DTMF digit duration is wrong

-DTMF inter-digit duration is too long/short

-the line is being seized too early before it is ready to receive DTMF

 

All basically meaning the CO is not able to detect the destination I am trying to reach and hence why CME does its bridging feature after the 10s timeout.

 

The default DTMF Duration/DTMF inter-digit (100ms/100ms) did not work so I hunted around for some recommened Australian settings.  I found one document which mentioned using 80ms/80ms but this did not work either.  I then increased the pre-dial-delay for the FXO port from 1 to 2s but this did not help either.  I noticed via a show voice port 3/0 that the companding type is set to u-law.  I know in Australia we use a-law but I believe this setting is only applicable for digital signalling (ISDN/Q.931)

 

I have also set the interface impedance to 220+820 | 120uF  (which I changed via setting to impedance complex1) and the tones to Aus specific (cptone AU)

 

One thing that confuses me a little is that the #show dialplan number command only matches my VoIP devcies and does not come back saying the POTS dial-peer would be used if I enter any number ( eg .T) other than the VoIP extensions .

 

Any comments/suggestions wold be appreciated

Regards

Ben

Attachments:
  • Paul Stewart  -  CCIE Security 7,570 posts since
    Jul 18, 2008
    Currently Being Moderated
    1. Oct 9, 2009 7:15 PM (in response to Ben)
    Re: FXO Problem

    I'm a little rusty on this, but did you wait for the interdigit timeout.  Since you are using "T", I think you need to wait something like 17 seconds after the last digit.

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  • Paul Stewart  -  CCIE Security 7,570 posts since
    Jul 18, 2008
    Currently Being Moderated
    3. Oct 9, 2009 8:19 PM (in response to Ben)
    Re: FXO Problem

    There could be something that is being misinterpreted as a supervisory disconnect from the PSTN.  You could disable it by going into the port and typing "timeout call-disconnect infinity".  The documentation indicates an incoming call is where this applies, but supervisory disconnect can happen on inbound or outbound calls.  At this point, I am just guessing someone else may have a better idea.  Also, if you have a test set to listen in on the call while it is happening, it might shed some light on what's going on.

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  • Conwyn 9,681 posts since
    Sep 10, 2008
    Currently Being Moderated
    6. Oct 12, 2009 4:42 AM (in response to Ben)
    Re: FXO Problem

    Hi Ben

     

    You might like to try this. STCAPP allows an analogue to be a virtual IP phone.

    Before you type in the ephone line you need the mac.

    show stcapp device summary will show you the virtual mac address. Leave in the existing FXO definitions except for the connection plar.

    You can forward the virtual phone to the real 7941 or  monitor the line using button 1:7941 2m80

     

    Referring back to PLAR see the example below and notice it uses button 1 & 2

     

    Regards Conwyn

     

    stcapp ccm-group 1
    stcapp
    !
    stcapp feature access-code

    voice-port 3/0
    timeouts ringing infinity
    caller-id enable

    dial-peer voice 1043 pots
    service stcapp
    port 3/0

    ephone-dn  80  dual-line
    number 8100
    label Analogue One
    name Analogue One
    ephone  7
    device-security-mode none
    mac-address 7954.7C2B.8100
    max-calls-per-button 2

    type anl
    button  1:80

     

    Alternatively PLAR solution

    router#show running-config
    voice-port 3/0
    connection plar-opx 1082
    dial-peer voice 82 pots
    destination-pattern 82
    port 3/0
    ephone-dn 10
    number 1010
    name manager
    ephone-dn 11
    number 1082
    name private-line
    trunk 82
    ephone 1
    button 1:10 2:11

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  • Conwyn 9,681 posts since
    Sep 10, 2008
    Currently Being Moderated
    9. Oct 12, 2009 11:20 AM (in response to Ben)
    Re: FXO Problem

    Hi Ben

     

    As you said PLAR forwards call to another telephone or group of telephones. Obviously you can only have one call coming in on the PSTN.

    One interesting thing with CME 4.3 / 7.0 was octoline. This allows eight lines on a button so you can give the receptionist a eight line phone so she can put calls on hold before transferring them. So you could have 8 FXO ports delivering to a receptionist. If you have 5 minutes spare try the virtual IP phone I was quite impressed.

     

    Regards Conwyn

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  • Conwyn 9,681 posts since
    Sep 10, 2008
    Currently Being Moderated
    11. Oct 12, 2009 11:39 PM (in response to Ben)
    Re: FXO Problem

    Hi Ben

     

    It is working on my Call Manager Express 4.3 quite happy. Have you tried the commands?

     

    Regards Conwyn

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  • jhubel 9 posts since
    Feb 17, 2009
    Currently Being Moderated
    13. Oct 27, 2009 1:26 PM (in response to Ben)
    Re: FXO Problem

    Ben -

     

    I just experienced a problem exactly like yours over the weekend.  The symptoms were:

    1) An outbound call through the FXO port would produce a quiet 'click', followed by direct loopback of the audio on the telephone.  This would last around 17 seconds, when the port would then release and the call drops.

    2) An inbound call through the FXO port would follow the plar statement on the voice-port, and ring the appropriate phone.  When picked up, the call would produce a connection, but no audio in either direction.

     

    As I began Googling the answer, I found this thread and was disappointed when no one had an answer.  While working on it again last night, I was able to solve the issue in my situation.  I moved the PSTN line to port 2 on my FXO card and modified my outgoing dial-peers (and moved my plar statement, too) and that solved it.  I now have normal functionality of the port.

     

    Another troubleshooting step you can perform is to plug a standard phone cable from your FXO port into an open FXS port.  When dialing a number which routes out the FXO port, you should then receive a second dial tone from the FXS port which subsequently goes off hook.

     

    Hope this helps you, but then again, I hope you don't have a hardware failure like me.

     

    Jeff

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