asterisk users - Sep 2007 - Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

An Asterisk extension calls an Alcatel extension via a
PRI link which rings 4 times for about 10-15 seconds
and then drops.
So if the Alcatel user doesn't answer within 10-15
seconds the call is aborted.
(A timeout is *not* specified in the Asterisk Dial
command.)
It seems however that either Asterisk or Alcatel drop
the call prematurely (it's more likely to be on the
Asterisk side).

What could I try?

The Asterisk log displays (* ext is 4053; Alcatel ext
is 5900):

-- Executing
Dial("SIP/4053-08311988","Zap/g1/5900||tTW") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/5900
-- Zap/2-1 is proceeding passing it to
SIP/4053-08311988
-- Zap/2-1 is ringing
-- Zap/2-1 is busy
-- Hungup 'Zap/2-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup("SIP/4053-08311988", "") in new
stack
== Spawn extension (from-internal, 5900, 4) exited
non-zero on 'SIP/4053-08311988'
-- Executing Macro("SIP/4053-08311988", "hangupcall")
in new stack
...etc...

The Alcatel board is configured as:

Interface Type + PRA2
CRC4 + YES
Retransmission Timer : 100
TEI Identity Check Timer : 100
Polling Timer : 1000
No. Of Retransmissions : 3
Max Frame Size (Bytes) : 260
Passive board + NO
SS7 signaling + NO

(I also tried to increase the above "Timer" values but
that did not change anything)

In Asterisk's /etc/zaptel.conf I have:

# TE120P (PRI):
span=1,1,0,ccs,hdb3,crc4

bchan=1-15
dchan=16
bchan=17-31

What could be the problem here?

Thanks



     
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Looks like the Alcatel is sending back a busy.  Check the value of 
HANGUPCAUSE with a Noop as the priority after the Dial.  You may also 
want to do a pri debug span X to see the actual Q.931 ISDN messages that 
are exchanged.

Vieri wrote:
> An Asterisk extension calls an Alcatel extension via a
> PRI link which rings 4 times for about 10-15 seconds
> and then drops.
> So if the Alcatel user doesn't answer within 10-15
> seconds the call is aborted.
> (A timeout is *not* specified in the Asterisk Dial
> command.)
> It seems however that either Asterisk or Alcatel drop
> the call prematurely (it's more likely to be on the
> Asterisk side).
> 
> What could I try?
> 
> The Asterisk log displays (* ext is 4053; Alcatel ext
> is 5900):
> 
> -- Executing
> Dial("SIP/4053-08311988","Zap/g1/5900||tTW") in new
> stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Called g1/5900
> -- Zap/2-1 is proceeding passing it to
> SIP/4053-08311988
> -- Zap/2-1 is ringing
> -- Zap/2-1 is busy
> -- Hungup 'Zap/2-1'
> == Everyone is busy/congested at this time (1:1/0/0)
> -- Executing Hangup("SIP/4053-08311988", "") in new
> stack
> == Spawn extension (from-internal, 5900, 4) exited
> non-zero on 'SIP/4053-08311988'
> -- Executing Macro("SIP/4053-08311988", "hangupcall")
> in new stack
> ...etc...
> 
> The Alcatel board is configured as:
> 
> Interface Type + PRA2
> CRC4 + YES
> Retransmission Timer : 100
> TEI Identity Check Timer : 100
> Polling Timer : 1000
> No. Of Retransmissions : 3
> Max Frame Size (Bytes) : 260
> Passive board + NO
> SS7 signaling + NO
> 
> (I also tried to increase the above "Timer" values but
> that did not change anything)
> 
> In Asterisk's /etc/zaptel.conf I have:
> 
> # TE120P (PRI):
> span=1,1,0,ccs,hdb3,crc4
> 
> bchan=1-15
> dchan=16
> bchan=17-31
> 
> What could be the problem here?
> 
> Thanks
> 
> 
> 
>      
____________________________________________________________________________________
> Luggage? GPS? Comic books? 
> Check out fitting gifts for grads at Yahoo! Search
>
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz
> 
> _______________________________________________
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