vicidial.org

Hi,

I am using astguiclient-2.0.5 trunk version. asterisk-1.4.21. And digium TE12XP card for pstn dialing.

I can able to dial absolutly fine. As card is configure properly. But i am getting the below error. Is this some serious error ?
-- Executing [failed@default:1] AGI("OutgoingSpoolFailed", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [failed@default:2] Dial("OutgoingSpoolFailed", "Zap/g1/failed|30|tTo") in new stack
[Jan 22 12:00:17] WARNING[21957]: channel.c:3009 ast_request: No translator path exists for channel type Zap (native 76) to 0
[Jan 22 12:00:17] WARNING[21957]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [failed@default:3] Hangup("OutgoingSpoolFailed", "") in new stack
== Spawn extension (default, failed, 3) exited non-zero on 'OutgoingSpoolFailed'
-- Executing [h@default:1] DeadAGI("OutgoingSpoolFailed", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----58-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing [h@default:2] Dial("OutgoingSpoolFailed", "Zap/g1/h|30|tTo") in new stack
[Jan 22 12:00:17] WARNING[21957]: channel.c:3009 ast_request: No translator path exists for channel type Zap (native 76) to 0
[Jan 22 12:00:17] WARNING[21957]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [h@default:3] Hangup("OutgoingSpoolFailed", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'OutgoingSpoolFailed'
== Manager 'sendcron' logged off from 122.165.48.212


What should i do to remove this error.

Thanks

Looks like something is wrong with either your telco circuit or your configuration on your Digium card.

What can be wrong with telco circuit. What should i confim with them for circuit..

Below are the configuration of my zaptel.

/etc/asterisk/zapata.conf


[channels]
sendcalleridafter=1
language=en
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
faxdetect=no
signalling=pri_cpe
switchtype=euroisdn
;channel => 1-23
channel => 1-15
channel => 17-31
;channel => 17-31


/etc/zaptel.conf

# Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER)
span=1,0,0,ccs,hdb3
#termtype: te
bchan=1-15,17-31
dchan=16

# Global data

loadzone = us
defaultzone = us


/etc/sysconfig/zaptel

MODULES="$MODULES wcte12xp" # TE120P - Single Span T1/E1 Card

If you find any misconfiguration than please tell me.

Thanks...

What country is this circuit in?

Have you received a cut-sheet from the carrier for this circuit?

This circuit is from INDIA. And what info do i need to get from provider. ? What is this cut-sheet circuit. ? They just told me that they are using Framing as ccs and Coding as hdb3. What else do i need to get from them.

Thanks

You should take a look at the sample zaptel.conf and zapata.conf files that are included with Asterisk, they should give you a better idea of how to configure those spans.