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

    FXO Problem

    Ben

      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

        • 1. Re: FXO Problem
          Paul Stewart  -  CCIE Security

          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.

          • 2. Re: FXO Problem
            Ben

            Hi Paul

             

            Thanks for your suggestion and v quick response; changing the destiination pattern has certainly helped as the pots dial-peer is now matched when I do a #show dialplan number command.  I played around with the DTMF digit duration/inter values again after this change but still no cigar

             

            Attached is the latest trace for reference.  Note the call seems to be bridged basically straight away (I get one ringing indication sequence on the VoIP handset) and then it is in an empty call (this is supported by the trace timing and there is no ~10s gap followed by the htsp_call_bridged invoked message. I'm not sure why the line is going back On hook so quickly.

             

            Regards

            Ben

            • 3. Re: FXO Problem
              Paul Stewart  -  CCIE Security

              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.

              • 4. Re: FXO Problem
                Ben

                Paul,

                 

                I tried your configuration suggestion, but was still unable to dial out

                I'm having some other un-related issues wth QoS on my setup relating to the IP blue softphones; I have recently purchased a couple of 7941G's that should shortly be arriving; I'm going reinvestigate when I have 'real' phones.

                Thanks for your thoughts

                • 5. Re: FXO Problem
                  Ben

                  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

                  • 6. Re: FXO Problem
                    Conwyn

                    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

                    • 7. Re: FXO Problem
                      Ben

                      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

                      • 8. Re: FXO Problem
                        Ben

                        Appologies Conwyn, it is getting late and I am starting to mistype things like people's names

                        • 9. Re: FXO Problem
                          Conwyn

                          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

                          • 10. Re: FXO Problem
                            Ben

                            Hi Conwyn

                             

                            I looked into implementing the virtual IP phone using STCAPP like you suggested, however this feature is only present on Call Manager 4.1 and higher (no CUCME support, see http://www.cisco.com/en/US/docs/ios/12_3t/12_3t14/feature/guide/gtstcapp.html#wpxref64786

                             

                            I'm going to ask the guy who sold me the 1760V & VIC-2FXO card whether he ever got it working, perhaps he was just using it as a gateway with CUCM and thus had no problems making/receiving calls under CUCME using the FXO port

                             

                            Regards

                            Ben

                            • 11. Re: FXO Problem
                              Conwyn

                              Hi Ben

                               

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

                               

                              Regards Conwyn

                              • 12. Re: FXO Problem
                                Ben

                                Hi Conwyn,

                                 

                                Yes that was the first thing I tried; and when the commands were not there I went hunting around trying to figure out why and found that cisco.com document.

                                So somewhere between CUCME 4.1 and CUCME 4.3 the feature was added in.

                                • 13. Re: FXO Problem
                                  jhubel

                                  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

                                  • 14. Re: FXO Problem
                                    Ben

                                    Jeff,

                                     

                                    Indeed it sounds like you had the same problem; congrats on figuring it out.

                                     

                                    I parked this issue as at the time as it appeared the solution required CME 4.3/7.0 using STCAPP.

                                     

                                    What version are you running?

                                     

                                    I don't have an FXS port in my rig, but it sounds like that it really helped you debug the problem and isolate what was going on

                                     

                                    Regards

                                    Ben

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