[Asterisk-Users] PRI configuration problem

Fernando Macías fmacias at validata.com.mx
Mon Dec 6 06:51:51 MST 2004


Shouldn't your spans be configured as cas,hdb3 rather than esf,b8sz?

Fernando


On Dec 6, 2004, at 12:33 AM, Andrew Aken wrote:

> We've been working for the past 2 weeks to get a new V400P working 
> with our PRIs from the telephone company. We're trying to get the 
> Asterisk server setup as a VoIP gateway for SIP and AIX. We can make 
> SIP-SIP calls, but all calls from or to the PRI fail. This is the 
> applicable entries from the Asterisk log (configuration files follow) 
> for a call coming from the PSTN on the PRI. I believe that the cause 
> of the error is related to the line, "Ring requested on unconfigured 
> channel 0/23 span 1". But as far as I can tell, the channels are all 
> configured.
>
> < Protocol Discriminator: Q.931 (8)  len=45
> < Call Ref: len= 2 (reference 1/0x1) (Originator)
> < Message type: SETUP (5)
> < [04 03 90 90 a2]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
> capability: 3.1kHz audio (16)
> <                              Ext: 1  Trans mode/rate: 64kbps, 
> circuit-mode (16)
> <                              Ext: 1  User information layer 1: u-Law 
> (34)
> < [18 03 a9 83 97]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
> Exclusive Dchan: 0
> <                        ChanSel: Reserved
> <                       Ext: 1  Coding: 0   Number Specified   Channel 
> Type: 3
> <                       Ext: 1  Channel: 23 ]
> < [1e 02 8a 01]
> < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
> (0) 0: 0   Location: Network beyond the interworking point (10)
> <                               Ext: 0  Progress Description: Call is 
> not end-to-end ISDN; further call progress information may be 
> available inband. (1) ]
> < [6c 0b 80 36 31 38 34 33 34 31 30 30 30]
> < Calling Number (len=13) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
> Unknown Number Plan (0)
> <                           Presentation: Presentation permitted, user 
> number not screened (0) '6184341000' ]
> < [70 0b a1 36 31 38 34 33 34 31 35 30 30]
> < Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6184341500' ]
> -- Making new call for cr 1
> -- Processing Q.931 Call Setup
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs0, Channel Identification)
> -- Processing IE 30 (cs0, Progress Indicator)
> -- Processing IE 108 (cs0, Calling Party Number)
> -- Processing IE 112 (cs0, Called Party Number)
> Dec  6 04:19:43 WARNING[4891]: Ring requested on unconfigured channel 
> 0/23 span 1
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, 
> peerstate Call Initiated
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 1/0x1) (Terminator)
> > Message type: RELEASE COMPLETE (90)
> > [08 02 81 ac]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
> Location: Private network serving the local user (1)
> >                  Ext: 1  Cause: Requested channel not available 
> (44), class = Network Congestion (2) ]
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
> NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
> ====================================
> Zaptel.conf
> -----------
> span=1,1,0,esf,b8zs
> span=2,2,0,esf,b8zs
> span=3,0,0,esf,b8zs
> span=4,0,0,esf,b8zs
> bchan=1-23
> dchan=24
> bchan=25-47
> dchan=48
> bchan=49-96
> loadzone = us
> defaultzone=us
> =====================================
> Zapata.conf
> -----------
> [trunkgroups]
> trunkgroup => 1,24,48
> spanmap => 1,1,1
> spanmap => 2,1,2
> spanmap => 3,1,3
> spanmap => 4,1,4
>
> [channels]
> group=1
> callgroup=1
> pickupgroup=1
> context=from-pstn
> switchtype=national
> signalling=pri_cpe
> channel => 1-23,25-47,49-96
> language=en
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> restrictcid=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> immediate=no
> callerid=asreceived
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=400
> ================================
> Extensions.conf
> ---------------
> [general]
> static=yes
> writeprotect=yes
>
> [from-pstn]
> exten => 6184341500,1,Dial(SIP/6184341500,20)
> exten => 6184341500,2,Voicemail2(u6184341500)
> exten => 6184341500,102,Voicemail2(b6184341500)
> exten => 6184341500,103,Hangup
> exten => 4341500,1,Dial(SIP/6184341500,20)
> exten => 4341500,2,Voicemail2(u6184341500)
> exten => 4341500,102,Voicemail2(b6184341500)
> exten => 4341500,103,Hangup
>
> [from-internal]
> exten => _NXXXXXX,1,Dial(Zap/g1/$(EXTEN))
> exten => _NXXXXXX,2,Congestion
> =======================================
> Sip.conf
> --------
> [6184341500]
> callerid="GlobalEyes" <6184341500>
> canreinvite=no
> context=from-internal
> dtmfmode=rfc2833
> host=dynamic
> mailbox=xxx
> nat=yes
> port=5060
> secret=xxx
> type=friend
> username=xxx
> allow=all
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list