[Asterisk-Users] Need help on * anf HFC.

[Ramon Roca]

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 *@home.
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 
'd804e3d3-299217a8 at 10.138.0.20' 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:200 at 10.138.0.20:5061>
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 <ramon.roca at guifi.net> 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.
>  
>