[cisco-voip] ISDN T1 voice calls hardware validation

PF pucknether at foril.com
Fri May 4 15:42:00 EDT 2018


Hi

Thanks a lot

Pat



  ----- Original Message ----- 
  From: Andrew Dorsett 
  To: PF 
  Cc: Ryan Huff ; cisco-voip voyp list 
  Sent: Friday, May 04, 2018 3:16 PM
  Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation


  Pat -
  I will agree that the ASR1k offers redundant RPs but by the time you configure it the way you want, you'd probably find dual ISR4k's to be the same cost.  Just a guess there as I haven't ran the numbers but I know the ASR comes at quite the premium over the ISR.



  The ISR4k offers an 8xT1 NIM and the ISR-4451 supports 5 NIMs (using SM-X NIM adapter) which gives you 40 T1 ports and 1200 E1 channels or 920 T1 channels in a single 2RU box.


  For the ASR to do voice decoding it would have to carry the TDM clock signal from the T1 SPA to the DSP SPA which doesn't happen.  On the old ISRG2's and older we carried the TDM clock across the backplane which meant you had a single clock domain inside the entire box.  The problem with that was you could encounter line slips if one of your circuits was not coming from the CO or from even the same provider in the same CO.  As a result, the new ISR4k does not carry the TDM clock outside of the NIM module and so the DSP resources all reside locally to the T1 cards themselves.  This means that you have to keep the same providers/CO on the same NIM but you can bring other providers/COs into other NIMs and have separate clocking domains.  This is huge for teleco redundancy across your PRIs.


  Andrew




  On Fri, May 4, 2018 at 2:47 PM, PF <pucknether at foril.com> wrote:

    ok

    I guess we will have to look for another product like the ISR4k

    the ASR1006 was very interresting because it was full redundant
    and we could put several 8xT1 card in it to have greater density

    the ISR4k seems to be limited to 3 cards and is not redundant.

    Pat

      ----- Original Message ----- 
      From: Andrew Dorsett 
      To: Ryan Huff 
      Cc: PF ; cisco-voip at puck.nether.net 
      Sent: Friday, May 04, 2018 12:45 PM
      Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation


      I replied privately but the SPA-DSP was built for SBC transcoding using CUBE between SIP or H323 trunks.  Unless something has changed recently it cannot be linked to the T1 SPA to terminate PRI Voice. 


      To terminate voice from a PRI you will need to use one of the ISR products and the relevant T1 module with DSPs. If you’re using the latest it would be an ISR4k with the NIM first mentioned and make sure the DSPs are onboard the NIM. 


      Andrew




      On Fri, May 4, 2018 at 12:32 PM Ryan Huff <ryanhuff at outlook.com> wrote:

        Not sure if DSP is onboard the SPA like the NIM; I’d have to look that one up ... you might try dsp services dspfarm under the voice-card configuration parameter. I’d also disable cdp on the serial interface. 


        Sent from my iPhone

        On May 4, 2018, at 12:18, PF <pucknether at foril.com> wrote:


          Hi

          I have tried several configurations

          Here is the relevent par of the actual config

          version 15.3
          boot system flash bootflash:asr1000rp2-adventerprisek9.03.10.07.S.153-3.S7-ext.bin
          !
          card type t1 0 2
          !
          multilink bundle-name authenticated
          isdn switch-type primary-5ess
          !
          voice-card 0/0
          !
          voice service pots
           supported-language FR
          !
          voice service voip
           clid network-provided
           allow-connections h323 to sip
           allow-connections sip to h323
           allow-connections sip to sip
           signaling forward unconditional
           sip
            bind control source-interface GigabitEthernet1/2/0
            bind media source-interface GigabitEthernet1/2/0
            ds0-num
            header-passing
            privacy id
            outbound-proxy ipv4:172.16.x.y
          !
          voice class codec 1
           codec preference 1 g711ulaw
          !
          controller T1 0/2/0
           framing esf
           clock source internal
           linecode b8zs
           cablelength long 0db
           pri-group timeslots 1-24
          !
          interface Service-Engine0/0/0
          !
          interface Serial0/2/0:23
           encapsulation hdlc
           isdn switch-type primary-5ess
           isdn negotiate-bchan
          !
          map-class dialer DOVtest
           dialer voice-call
          !
          dspfarm profile 1 transcode universal  
           rsvp
           shutdown
          !
          dial-peer voice 1 voip
           destination-pattern *.
           signaling forward unconditional
           session protocol sipv2
           session target sip-server
           voice-class codec 1  
          !
          dial-peer voice 2 pots
           destination-pattern xxxxxxx
           incoming called-number xxxxxxx
           direct-inward-dial
          !
          !
          sip-ua 
           credentials username asr1006 password 7 xxxxxxxx realm yyyyy
           retry invite 3
           retry bye 3
           retry cancel 3
           timers trying 1000
           timers register 100
           sip-server ipv4:172.16.x.y
          !
          !
          Pat


            ----- Original Message ----- 
            From: Ryan Huff 
            To: Patrick Fortin 
            Cc: cisco-voip at puck.nether.net 
            Sent: Friday, May 04, 2018 11:43 AM
            Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation


            I assume you have the card type specified and controller interface configured with timeslots? 


            Can you send the running-config and code version?


            Sent from my iPhone

            On May 4, 2018, at 11:35, Patrick Fortin <pfortin at royaume.com> wrote:


              Hi

              I get this error

               **ERROR**: call_incoming: Received a call id 0x2F with a bad bearercap from xxxxxxxxxx on b channel 1

              seems like the dsp are not associated with the t1 card

              in the config we don't have acces to the following :
              isdn incoming-voice voice
              which should go on the Serial interface. 

              we also don't have the "port" command that should go in the dial-peer voice section

              any ideas ?

              Thanks

              Pat
                ----- Original Message ----- 
                From: Ryan Huff 
                To: PF 
                Cc: cisco-voip at puck.nether.net 
                Sent: Friday, May 04, 2018 11:11 AM
                Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation


                That correct! I misread! Yeah the shared port adapter T1 should work.  



                Sent from my iPhone

                On May 4, 2018, at 10:36, PF <pucknether at foril.com> wrote:


                  Hi

                  Thanks for your help

                  But there are no NIM slot in the ASR1006 chassis

                  NIM is for ASR1001-X I think

                  Pat

                    ----- Original Message ----- 
                    From: Ryan Huff 
                    To: PF 
                    Cc: cisco-voip at puck.nether.net 
                    Sent: Friday, May 04, 2018 9:56 AM
                    Subject: Re: [cisco-voip] ISDN T1 voice calls hardware validation


                    For isdn voice you’ll need a NIM-8MFT-T1/E1


                    Sent from my iPhone

                    On May 4, 2018, at 09:31, PF <pucknether at foril.com> wrote:


                      Hi
                      Can someone help us validate if we can use this hardware to receive voice calls from a isdn T1 (23B+D) and send them in SIP to a softswitch and vice-versa

                      ASR1006
                      SPA-8XCHT1/E1
                      SPA-DSP
                      ASR1000-RP2
                      ASR1000-ESP40

                      in short can it be used to build a voip gateway like an audiocode mediant or a patton smartnode

                      Thanks

                      Pat

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