Hey Julian, thanks! It really make a difference. Thanks for pointing me
to this. Stupid newbie mistake.
Yes, I'm using AMP, it was bundled with [EMAIL PROTECTED]
Now I'm not longer getting the all-the-circuits-are-busy-now, but still
doesn't dial out, now I'm getting the congestion tone.
Maybe I'm loading the zaphfc with wrong parameters for Spanish ISDN?
I'm just using a regular ISDN at home, and plugged the RJ45 cable at the
same port where was the Euromix RDSI phone.
Here it is the current * console while dialing out:
Mar 6 22:44:58 DEBUG[3700]: Setting NAT on RTP to 0
Mar 6 22:44:58 DEBUG[3700]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 101: Found
Mar 6 22:44:58 DEBUG[3700]: Setting NAT on RTP to 0
Mar 6 22:44:58 DEBUG[3700]: Check for res for 200
Mar 6 22:44:58 DEBUG[3700]: Call from user '200' is 1 out of 0
Mar 6 22:44:58 DEBUG[3700]: build_route: Contact hop: Roser Roca
sip:[EMAIL PROTECTED]:5061
Mar 6 22:44:58 VERBOSE[3700]: -- Executing Macro(SIP/200-bd90,
dialout-default|9639712471) in new stack
Mar 6 22:44:58 WARNING[3700]: ast_yyerror(): syntax error: parse error;
Input:
fooEl Serrat = foo
^
^
Mar 6 22:44:58 DEBUG[3700]: Expression is 'fooEl'
Mar 6 22:44:58 VERBOSE[3700]: -- Executing GotoIf(SIP/200-bd90,
fooEl?4) in new stack
Mar 6 22:44:58 DEBUG[3700]: Not taking any branch
Mar 6 22:44:58 VERBOSE[3700]: -- Executing
SetCallerID(SIP/200-bd90, El Serrat) in new stack
Mar 6 22:44:58 VERBOSE[3700]: -- Executing Goto(SIP/200-bd90,
6) in new stack
Mar 6 22:44:58 VERBOSE[3700]: -- Goto (macro-dialout-default,s,6)
Mar 6 22:44:58 VERBOSE[3700]: -- Executing Dial(SIP/200-bd90,
ZAP/g0/9639712471) in new stack
Mar 6 22:44:58 VERBOSE[3700]: -- Called g0/9639712471
Mar 6 22:45:02 VERBOSE[3700]: -- Channel 0/1, span 1 got hangup
Mar 6 22:45:02 DEBUG[3700]: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Mar 6 22:45:02 DEBUG[3700]: Hangup: channel: 1 index = 0, normal = 15,
callwait = -1, thirdcall = -1
Mar 6 22:45:02 DEBUG[3700]: Already hungup... Calling hangup once, and
clearing call
Mar 6 22:45:02 DEBUG[3700]: disabled echo cancellation on channel 1
Mar 6 22:45:02 DEBUG[3700]: Set option TDD MODE, value: OFF(0) on Zap/1-1
Mar 6 22:45:02 DEBUG[3700]: Updated conferencing on 1, with 0
conference users
Mar 6 22:45:02 DEBUG[3700]: Set option AUDIO MODE, value: OFF(0) on Zap/1-1
Mar 6 22:45:02 DEBUG[3700]: disabled echo cancellation on channel 1
Mar 6 22:45:02 VERBOSE[3700]: -- Hungup 'Zap/1-1'
Mar 6 22:45:02 VERBOSE[3700]: == No one is available to answer at
this time
Mar 6 22:45:02 DEBUG[3700]: Exiting with DIALSTATUS=NOANSWER.
Mar 6 22:45:02 VERBOSE[3700]: -- Executing
Congestion(SIP/200-bd90, ) in new stack
Mar 6 22:45:02 VERBOSE[3700]: == Spawn extension
(macro-dialout-default, s, 7) exited non-zero on 'SIP/200-bd90' in macro
'dialout-default'
Mar 6 22:45:02 VERBOSE[3700]: == Spawn extension (from-internal,
9639712471, 1) exited non-zero on 'SIP/200-bd90'
Mar 6 22:45:02 VERBOSE[3700]: -- Executing Macro(SIP/200-bd90,
hangupcall) in new stack
En/na Julian J. M. ha escrit:
Hello,
I don't know if your zaptel.conf and zapata.conf setup regarding your
isdn is correct, but if you use the default AMP setup, you need to
assign your channels to group 0 for dialing out, and assign it to
context from-pstn if you want to receive calls.
group = 0
context=from-pstn
channel = 1-2
BTW, i'm from Spaintoo, and I'm really interested in knowing if your
setup works ;)
On Sun, 06 Mar 2005 21:41:17 +0100, Ramon Roca [EMAIL PROTECTED] wrote:
[channels]
group = 1
context=outbound-trunks
channel = 1-2
Mar 6 21:40:01 VERBOSE[21452]: -- Executing Dial(SIP/200-1cf6,
ZAP/g0/9639712471) in new stack
g0 means channel group 0, and you had group 1
Julian.
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