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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