Need help with D110P - Non PRI US t1 "AT&T Style"?

Discussion in 'General' started by amartin, Mar 29, 2009.

  1. amartin

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    Hi all I need a little help with my T1 card. Outgoing calls have no audio, but incoming calls work great! I'm using elastix 1.3-2 here are my files and a screen shot of zttool when I'm making an outgoing call notice RX is 0000.

    [​IMG]

    /etc/asterisk/zapata.conf
    Code:
    [trunkgroups]
    
    [channels]
    context=from-pstn
    signalling=em_w
    rxwink=300              ; Atlas seems to use long (250ms) winks
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    usecallingpres=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    canpark=yes
    cancallforward=yes
    callreturn=yes
    echocancel=yes
    echocancelwhenbridged=no
    faxdetect=incoming
    echotraining=800
    rxgain=0.0
    txgain=0.0
    callgroup=1
    pickupgroup=1
    
    ;Uncomment these lines if you have problems with the disconection of your analog lines
    ;busydetect=yes
    ;busycount=3
    
    
    immediate=no
    
    #include zapata_additional.conf
    #include zapata-channels.conf
    
    /etc/zaptel.conf
    Code:
    # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
    # Zaptel Configuration File
    #
    # This file is parsed by the Zaptel Configurator, ztcfg
    #
    
    # It must be in the module loading order
    
    
    # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) B8ZS/ESF
    span=1,1,0,esf,b8zs
    # termtype: te
    e&m=1-24
    
    # Global data
    
    loadzone        = us
    defaultzone     = us
    
    /etc/asterisk/zapata-channels.conf
    Code:
    ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
    ; Zaptel Channels Configurations (zapata.conf)
    ;
    ; This is not intended to be a complete zapata.conf. Rather, it is intended
    ; to be #include-d by /etc/zapata.conf that will include the global settings
    ;
    
    ; Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) B8ZS/ESF
    group=0,11
    context=from-pstn
    switchtype = national
    signalling = e&m
    channel => 1-23
    group=
    context=default
    
    
     
  2. dicko

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    Re:Need help with D110P - Non PRI US t1

    but also, do you have ulaw specifically enabled in the extensions?

    rasterisk -x "sip show peer <a real one>" and look at "codec order"
    and yes RxA and RxB should "wink" 1 then change to 1 on successful dial
     
  3. amartin

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    Re:Need help with D110P - Non PRI US t1

    So I changed my zapata-channels.conf

    ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
    ; Zaptel Channels Configurations (zapata.conf)
    ;
    ; This is not intended to be a complete zapata.conf. Rather, it is intended
    ; to be #include-d by /etc/zapata.conf that will include the global settings
    ;

    ; Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) B8ZS/ESF
    group=0
    context=from-pstn
    signalling = em_w
    channel => 1-24

    I was using the defaults of what was there before. I read your other thread, you helping the guy who wasn't sure if he had a PRI or E&M Wink T1. My trunk is configured as Zap/G0. If I add my sip provider to the available outgoing routes it dials correctly. Here is the info you asked for.

    Code:
    [root@elastix ~]# rasterisk -x "sip show peer 101" and look at "codec order"
    
    
      * Name       : 101
      Secret       : <Set>
      MD5Secret    : <Not set>
      Context      : from-internal
      Subscr.Cont. : <Not set>
      Language     :
      AMA flags    : Unknown
      Transfer mode: open
      CallingPres  : Presentation Allowed, Not Screened
      Callgroup    :
      Pickupgroup  :
      Mailbox      : 101@default
      VM Extension : *97
      LastMsgsSent : 0/0
      Call limit   : 50
      Dynamic      : Yes
      Callerid     : "device" <101>
      MaxCallBR    : 384 kbps
      Expire       : 3410
      Insecure     : no
      Nat          : Always
      ACL          : No
      T38 pt UDPTL : No
      CanReinvite  : No
      PromiscRedir : No
      User=Phone   : No
      Video Support: No
      Trust RPID   : No
      Send RPID    : No
      Subscriptions: Yes
      Overlap dial : Yes
      DTMFmode     : rfc2833
      LastMsg      : 0
      ToHost       :
      Addr->IP     : 10.0.40.67 Port 5060
      Defaddr->IP  : 0.0.0.0 Port 5060
      Def. Username: 101
      SIP Options  : (none)
      Codecs       : 0xc (ulaw|alaw)
      Codec Order  : (ulaw:20,alaw:20)
      Auto-Framing:  No
      Status       : OK (21 ms)
      Useragent    :
      Reg. Contact : sip:101@10.0.40.67:5060
    
    
    
     
  4. amartin

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    Re:Need help with D110P - Non PRI US t1

    On the sip comment, it dials correctly via the sip trunk, adding the sip provider doesn't make the T1 outgoing work correctly.
     
  5. dicko

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    Re:Need help with D110P - Non PRI US t1

    do you have callerID on the trunk, (then signalling=feat_d probably)
    Does the telco send "dialtone" on receipt of your trunk wink? (sometimes they don't so you might need immediate=yes.)
    do you see the channel "wink" (in zttool) on outbound dialing, (that would be on channel 24 if using ZAP/G0)
    Do you have DID's on the trunk or is it just one number?
    Did you stop and start asterisk after each change to zapata.conf?

    I would remove the autogenerated file and do it by hand in zapata.conf
    with reference to:
    http://www.voip-info.org/wiki-Asterisk+ ... apata.conf

    There are so many variables, you should have very explicit information from the telco to speed the process)
     
  6. amartin

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    Re:Need help with D110P - Non PRI US t1

    I have a new development, I changed my settings to em from em_w and now I can call out correctly but get no audio when I call into the system.
     
  7. dicko

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    Re:Need help with D110P - Non PRI US t1

    then back to em_w and immediate=yes

    My guess is that the telco is not sending you dialtone on wink, you should ask them to change that for reliability and trunk symmetry, you didn't reply to DID, callerID questions.
     
  8. amartin

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    Re:Need help with D110P - Non PRI US t1

    NICE! Works great!

    Ok, one last question, how do I make it answer calls on channels 1 and up.
    Out going calls out on channel 24 and down?

    I shall donate to your cause elastix has and will save our company lots of $$.
     
  9. dicko

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    Re:Need help with D110P - Non PRI US t1

    You're welcome and Elastix appreciates your generosity.

    Inbound the Telco is responsible for hunt behavior, normally 1,2,3,4 . .

    Simply by creating and using zap/G0 (instead of ZAP/g0) in your outbound routes, you will use 24,23,22 . . .

    For T1 "tuning" read my posts on milliwatt testing, it will almost certainly improve your users experience and echo reduction.
     
  10. amartin

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    Re:Need help with D110P - Non PRI US t1

    So, I ran into a little bump. While trouble shooting I accidentally changed G0 to g0. Now, it's back to not properly dialing out again. If I change it back to g0 it works correctly. I tried going to g1 and G1 and changing the group=1 and I get the same result. This is probably something simple.

    edit: Forget it I figured it out, the channel wasn't active
     
  11. dicko

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    Re:Need help with D110P - Non PRI US t1

    Did you build the trunk from within FreePBX
    "Trunks-> add ZAP trunk-> Zap Identifier (trunk name)=G0"
    ?
    Does the Telco honor all 24 ds0's (it is not a "fractional" T1, where only ds0 1-<limit> are honored), if so channel=1-<limit>

    To trouble shoot you can access the individual DS0's (channels) from Asterisk CLI with
    originate ZAP/n/13235551212 extension 101
    where n = 1-24 a call will be placed to 13235551212 from ext. 101 on channel n
     
  12. vloose

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    Re:Need help with D110P - Non PRI US t1

    Where can I find your post on "milliwatt testing." Sorry if my searching skills are lacking a bit, but I would love to read up because of some echo problems I am running into. I just can't find the post.
     
  13. dicko

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