IIUC Notes – Voice Ports and Dial Peers

 

More of my IIUC study notes.  As always, feel free to correct.  I really need to have a real post, don't I?

show voice port summary

  • Shows the voice ports available for use

  • An ephone-dn shows up as efxs, so all these are ephone-dns.
  • Channels are numbered 0-23; timeslots are numbered 1-24

FXS Ports

  • Connect to end stations like analog phones and fax machines
  • Signaling
    • Ground start: New connections started by grounding wires
      • Typically used when tied to PBXes
    • Loop start:  New connections started by sending DC voltage
      • Default
      • Typically used when connecting to analog devices
  • Call progress tones
    • Audible tones to let the user know the status of a call
      • Dial tone, busy, call waiting, etc.
      • Different in each geographical area
  • Caller ID
    • Identifies the name and number that calls on this line should appear

R1(config)#voice-port 0/0/0
R1(config-voiceport)#signal loopStart <- Use loopstart signaling
R1(config-voiceport)#cptone PE <- Uses CP tones from Peru
R1(config-voiceport)#station-id name Corporate Fax
R1(config-voiceport)#station-id number 5551212

FXO Ports

  • Connects to CO or PBX
  • A lot of the same configurations as FXS ports
  • Two additional to discuss
    • dialt-type:  DTMF or pulse dialing
    • ring:  The number of rings to wait before answering; usually 1
      • Think of allowing a home user to answer the phone before the fax machine picks up

R1(config)#voice-port Reverse Phone 0/0/1
R1(config-voiceport)#dial-type dtmf <- touch tone
R1(config-voiceport)#ring 3 <- wait 3 rings before answering

Digital Voice Ports

  • Unlike analog voice ports, digital voice ports must be configured to function with the network to which they are attached.
  • Voice and WAN interface cards (VWICs) provide digital voice port
  • show controllers t1
  • Framing:  defines how to format the frames
    • SF or ESF
  • Line coding:  encodes the signal in a way to maintain sychronization
    • AMI or B8ZS
  • Clock source:  defines who dictates the clocking
  • Signaling:  channel signaling
    • CAS:  use ds0-group
      • Ports show up as 0/0:1, where 0/0 is the physical port and 1 is the ds0 group
    • CCS:  use pri-group
      • Ports shows up as 0/0:23, where 0/0 is the physical port and 23 is the signaling channel (16 in E1)

R1(config)#isdn switch-type primary-5ess <- If using CCS
R1(config)#controller t1 0/0
R1(config-controller)#framing esf
R1(config-controller)#linecode b8zs
R1(config-controller)#clock source line <- get clocking from provider
For CAS:
R1(config-controller)#ds0-group 1 timeslots 1-24 type fxo-loop-start <- Using FXO loopstart signaling
-or-
For CCS:
R1(config-controller)#pri-group 1 timeslots 1-24 <- assumes signaling from CCS and ISDN switch-type

Dial Peers

  • "Routing" for phone numbers
  • Tells a voice gateway where to send calls based on dialed number
  • Two types dial peers
    • POTS:  Traditional connections like T1 and analog phone lines
    • VOIP:  Connections to an IP address
  • show dial-peer voice summary

R1(config)#dial-peer voice 1101 pots
R1(config-dial-peer)#destination-pattern 1101  <- This number…
R1(config-dial-peer)#port 0/0/0  <-  …is on this FXS port.

R1(config)#dial-peer voice 1102 pots
R1(config-dial-peer)#destination-pattern 1102  <- This number…
R1(config-dial-peer)#port 1/0:23  <-  …is on this T1 PRI port.

R1(config)#dial-peer voice 1103 voip
R1(config-dial-peer)#destination-pattern 1103  <- This number…
R1(config-dial-peer)#session target ipv4:10.10.10.1  <- …is at this IP address…
R1(config-dial-peer)#codec g711ulaw  <- …and use this codec when you get there.

Aaron Conaway

I shake my head around sometimes and see what falls out. That's what lands on these pages.

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7 comments for “IIUC Notes – Voice Ports and Dial Peers

  1. October 5, 2010 at 1:35 pm

    Why do you have both voip and pots dial-peers? On our gateway router (I'm not familiar with Cisco Voice) we have multiple pots dial-peers and a single voip dial-peer.

    Do dial-peers function both directions? (From inside to out, and outside to in)

  2. October 5, 2010 at 2:09 pm

    Hey, Robert.  If the session target of a dial-peer is a physical port (like an FXS, FXO, T1, etc.), yo use pots.  If the target is another voice gateway (like a router or UCM), then you use voip.

    Dial-peers don't work in both directions.  They only tell the call processor where to send the call.  If I called you from my CME that had a dial-peer, your phone would ring (if everything else was set up correctly).  If you called me from your CME without one, the call wouldn't make it.

  3. SNP
    July 31, 2011 at 9:55 am

    Aaron, I have a few gaps in my basic understanding of this material, can you help out? First, in response to your comment where you say dial-peers don’t work in both directions, are you talking about something else than outbound/inbound dial peers? I read that both must be configured for a call to go through.

    Okay, this is kind of embarrassing but what is the difference between a digital voice port and a VoIP port? Aren’t they both digital, and can’t they both support non-voice data as well? To which one would you connect something like a T3?

  4. July 31, 2011 at 11:15 am

    Hey, SNP.

    My comment was scoped more toward finding the destination of a call rather than the concept of inbound/outbound peers. I’m no voice guy (thank God!), so I’ll just leave it there to avoid embarrassing myself. 🙂

    A digital voice port is a physical port on a router. Think an ISDN T1 PRI module for your 3800 where a line from your provider is physically connected to your router. That’s a digital signal to your provider for carrying voice and sometimes data. A VoIP port (or peer) is a way to connect voice streams over IP. This uses other physical media (like your corporate network) to peer voice devices. I’ve not heard of using VoIP peers to send data, but I’m not even close to the authority on that. If you had digital voice coming to you via a T3, you would be using a digital voice port (as opposed to a VoIP port). Anything beyond that is beyond me.

    Hope that helps.

  5. SNP
    July 31, 2011 at 12:01 pm

    Ok thanks Aaron, that’s what I was looking for. I’m going into Voice/Media communications but there’s a lot of ground to cover and I have an uneven understanding of the landscape, and the fact that it’s always changing (such as the convergence of voice with video, other types of media, and of course “data”) isn’t helping. Is it accurate to say that in modern deployments voice gateways are acting only as backup mechanisms, and when working properly all inter-site voice/media communication occurs through IP over WAN-type connections?

  6. Joe
    May 8, 2014 at 12:35 pm

    I can’t thank you enough for what you wrote for the voice. It provided some encouragement, but I am still stuck with the lab. I am trying the IP telephony using call manager express lab 5-3.

  7. Joe Sullivan
    May 8, 2014 at 12:47 pm

    In your dial-per voice you listed 1/0:23 (T1 port).

    My T1 is serial 0/2/0 but the dial-peer voice 2 pots with port 0/2/0 is not available. Based on my device hardware below can you shine light on this? I put the T1 card in slot 0/2/0 in my 2811 router.

    Also for controller t-1 it is not available either, instead i type: UnityRtR(config-if)#service-module t1 ? (options below).

    WIC Slot 2:
    FT1 PMC4351
    Hardware Revision : 1.2
    Product (FRU) Number : WIC-1DSU-T1-V2

    interface Serial0/2/0
    no ip address
    service-module t1 timeslots 1-12

    UnityRtR(config-if)#service-module t1 ?
    cablelength Specify the cablelength
    clock Service module clock source
    data-coding Service module DCE line coding
    fdl Specify the FDL standard for a DS1 data link
    framing Framing format
    lbo Line Build Out format
    linecode Line coding format
    remote-alarm-enable Remote/Yellow alarm enable
    remote-loopback Remote Loopback/RL disable
    timeslots Fraction timeslots

    UnityRtR#sh service-module serial 0/2/0
    Interface Serial0/2/0
    Module type is T1/fractional
    Hardware revision is 1.2, Software revision is 20130214,
    Image checksum is 0x4347ED, Protocol revision is 0.1
    Receiver has no alarms.
    Framing is ESF, Line Code is B8ZS, Current clock source is line,
    Fraction has 12 timeslots (64 Kbits/sec each), Net bandwidth is 768 Kbits/sec.
    Last module self-test (done at startup): Passed
    Last clearing of alarm counters 01:09:13
    loss of signal : 0,
    loss of frame : 0,
    AIS alarm : 0,
    Remote alarm : 1, last occurred 01:08:35
    Module access errors : 0,
    Total Data (last 4 15 minute intervals):
    0 Line Code Violations, 0 Path Code Violations
    0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
    0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
    Data in current interval (549 seconds elapsed):
    0 Line Code Violations, 0 Path Code Violations
    0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
    0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
    UnityRtR#

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