Lee, John (Sydney)
2008-Feb-29 07:30 UTC
[asterisk-users] Can call in but cannot call out (CHANUNAVAIL): TE410 + Asterisk 1.4.13 + Zaptel 1.4.6 + Libpri 1.4.2
I encountered this strange problem which is I can call into Asterisk box
but I cannot call out.
I was successful before using exactly the same euroISDN line but with
TE110 and different versions of Asterisk.
This time, I am using:
. TE410
. Asterisk 1.4.13
. Zaptel 1.4.6
. Libpri 1.4.2
1) I put the following into extensions.conf to get to the outside line
exten => 0,1,Dial(Zap/1)
2) When I hit '0' on the phone, it came back like this on the console:
[Feb 29 18:15:40] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
rxwink
== Parsing '/etc/asterisk/users.conf': Found
-- Executing [0 at default:1] Dial("SIP/5166-b7b0caa0",
"Zap/1") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1CLI>
-- Zap/1-1 is proceeding passing it to SIP/5166-b7b0caa0
-- Channel 0/1, span 1 got hangup request, cause 31
[Feb 29 18:16:12] WARNING[7365]: app_dial.c:746 wait_for_answer: Unable
to forward voice or dtmf
-- Hungup 'Zap/1-1'
[Feb 29 18:16:12] NOTICE[7365]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/1-1' not posted
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/5166-b7b0caa0' status is
'CHANUNAVAIL'
3) When I did "reload chan_zap.so"
-- Reloading module 'chan_zap.so' (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
switchtype
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
pridialplan
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
prilocaldialplan
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
overlapdial
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
priindication
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
signalling
-- Reconfigured channel 1, ISDN PRI signalling
[...]
-- Reconfigured channel 31, ISDN PRI signalling
[Feb 29 18:24:31] WARNING[7146]: chan_zap.c:11120 process_zap: Ignoring
rxwink
== Parsing '/etc/asterisk/users.conf': Found
3) Something dodgy when I did a # cat /proc/zaptel/1
It complains that the channels are all "In Use" and I am only using
span
1
Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 ClockSour
ce
1 TE4/0/1/1 Clear (In use)
2 TE4/0/1/2 Clear (In use)
[...]
15 TE4/0/1/15 Clear (In use)
16 TE4/0/1/16 HDLCFCS (In use)
17 TE4/0/1/17 Clear (In use)
[...]
Hi , i've a problem with the dispatcher module of Openser, for the load
balancing for asterisk
The schema of the network is this :
Firewall (public ip:
199.199.199.199:5060)
|
|
-------------------
Pc1
pc2
Pc1: 192.168.0.1:5060 Asterisk
Pc2:192.168.0.3:5060 Asterisk
Pc1:192.168.0.2:5060 Openser
The ip public is natted with Openser
The dispatcher.list is:
1 sip:192.168.0.1:5060
1 sip:192.168.0.3:5060
Openser.cfg:
route{
if ((method=="INVITE") ){
ds_select_dst("1","4"); # 4 = round-robin
t_relay();
exit();
}
}
In asterisk for test there is an ivr ,and when I make a call from an other
site to this public ip, ast answer but after few seconds I see from the log
of the pc01 the the call hang but from the remote asterisk (the one where I
make the call) the call remain up.
The configuration of openser is correct ?
thanks
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