[SOLVED] * => PRI Com Problem (everyone busy/congested)


#1

I read through every forum and help file i could find to figure out what i did wrong, but nothing. (even though i did learn alot)

I know the switch is programmed correctly because it currently speaks with my 3com modem bay over the PRI.

I have the Dchannels and Bchannels up and teh restart signals are send/recieveing on both ends just fine…

I have the debug stuff running…so here is what i get…

    -- Executing Dial("SIP/phone1-d343", "Zap/g1/97522033|300|r") in new stack
-- Making new call for cr 32777
    -- Requested transfer capability: 0x00 - SPEECH
> Protocol Discriminator: Q.931 (8)  len=43
> Call Ref: len= 2 (reference 9/0x9) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a2]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>                              Ext: 1  User information layer 1: u-Law (34)
> [18 04 e9 81 83 81]
> Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0
>                        ChanSel: Reserved
>                       Ext: 1  DS1 Identifier: 1  
>                       Ext: 1  Coding: 0   Number Specified   Channel Type: 3
>                       Ext: 1  Channel: 1 ]
> [1e 02 80 83]
> Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: User (0)
>                               Ext: 1  Progress Description: Calling equipment is non-ISDN. (3) ]
> [28 05 b1 4d 69 6b 65]
> Display (len= 5) Charset: 31 [ Mike ]
> [6c 03 21 81 31]
> Calling Number (len= 5) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>                           Presentation: Presentation permitted, user number passed network screening (1) '1' ]
> [70 09 a1 39 37 35 32 32 30 33 33]
> Called Number (len=11) [ Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '97522033' ]
    -- Called g1/97522033
< Protocol Discriminator: Q.931 (8)  len=10
< Call Ref: len= 2 (reference 9/0x9) (Terminator)
< Message type: RELEASE COMPLETE (90)
< [08 03 80 e4 18]
< Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: User (0)
<                  Ext: 1  Cause: Unknown (100), class = Protocol Error (6) ]
<              Cause data 1: 18 (24)
-- Processing IE 8 (cs0, Cause)
    -- Channel 1/1, span 1 got hangup
    -- Channel 1/1, span 1 received AOC-E charging 0 units
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
    -- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)
Jun 12 06:36:44 WARNING[12008]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'sip'

Here are my config files:


"zapata.conf"

[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
;    group => <trunkgroup>,<dchannel>[,<backup1>...]
;
;        trunkgroup  is the numerical trunk group to create
;        dchannel    is the zap channel which will have the
;                    d-channel for the trunk.
;        backup1     is an optional list of backup d-channels.
;
;
trunkgroup => 1,24
trunkgroup => 2,48
trunkgroup => 3,72
trunkgroup => 4,96
;
; Spanmap: Associates a span with a trunk group
;        spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]
;
;        zapspan     is the zap span number to associate
;        trunkgroup  is the trunkgroup (specified above) for the mapping
;        logicalspan is the logical span number within the trunk group to use.
;                    if unspecified, no logical span number is used.
;
spanmap => 1,1,1
spanmap => 2,2,2
spanmap => 3,3,3
spanmap => 4,4,4


[channels]
context=default
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
immediate=no
;busydetect=no
;callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived

language=en


switchtype=national
signalling=pri_cpe
group=1
;resetinterval = 3600
priindication = outofband
channel => 1-23
channel => 25-47
channel => 49-71
channel => 73-95
"extensions.conf" 20L, 451C written
[root@AsteriskDev asterisk]# vi extensions.conf 

[default]
exten => 1,1,Dial(SIP/phone1,20,tr)
exten => 2,1,Dial(SIP/phone2,20,tr)
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN},300,r)
exten => _9XNXXNXXXXXX,1,Dial(Zap/g1/${EXTEN},300,r)


[sip]
exten => 1,1,Dial(SIP/phone1,20,tr)
exten => 2,1,Dial(SIP/phone2,20,tr)
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)
exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN},300,r)
exten => _9XNXXNXXXXXX,1,Dial(Zap/g1/${EXTEN},300,r)



[qwest]

its just Linux redhat with a 4port t1 tor2 card.
im trying to get it to talk to my Brooktrout ns300 switch (does call termination for my modem bay over pri (so i know taht side works)).

TIA for what ever help you can give me
ThreeM


#2

had to remove trunkgroup and span map stuff from config file cause i wasnt using NFAS