[Asterisk-Users] Asterisk with HT 488 FXO
Pasqualotto Enrico
pasqu at linux.it
Tue Feb 28 12:09:11 MST 2006
Soner Tari wrote:
> Hi Pasqualotto,
>
> Actually, I've seen your post on Asterisk-Users list yesterday, but I
could
> not understand back then. Now, I've checked your sip configuration
again, I
> think you make a mistake in "type" of sip account. I use "friend" not
> "peer". I am not sure though.
Ok, thanks, now with my new "type" the call from FXO (300) are correctly
forwarded to my extension (204) after n second.
Now I have another problem: I want that the calls from 300 to 204 are
redirected to my ring-group.
With A at H & Inbound routing I have add these lines in extension.conf:
-------------- cut ---------------------------
[ext-did]
include => ext-did-custom
exten => s/204,1,SetVar(FROM_DID=s/204)
exten => s/204,2,Goto(ext-group,1,1)
exten => _X./204,1,Goto(s/204)
[ext-group]
include => ext-group-custom
exten => 1,1,Macro(rg-group,ringall,60,,201-202-203-204-205-206)
exten => 1,2,Goto(ext-group,1,1) ; jump
-------------- cut -------------------------------------
The calls from context "from-pstn" (SIP account) is also redirected to
ring-group and these work.
I found this in Asterisk CLI:
-- Executing Macro("SIP/300-3bb9", "exten-vm|novm|204") in new stack
-- Executing Macro("SIP/300-3bb9", "user-callerid") in new stack
-- Executing DBget("SIP/300-3bb9", "AMPUSER=DEVICE/300/user") in
new stack
-- DBget: varname=AMPUSER, family=DEVICE, key=300/user
-- DBget: set variable AMPUSER to 300
-- Executing DBget("SIP/300-3bb9",
"AMPUSERCIDNAME=AMPUSER/300/cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=300/cidname
-- DBget: set variable AMPUSERCIDNAME to ht488
-- Executing GotoIf("SIP/300-3bb9", "0?5") in new stack
-- Executing SetCallerID("SIP/300-3bb9", ""ht488" <300>") in new stack
-- Executing NoOp("SIP/300-3bb9", "Using CallerID "ht488" <300>")
in new stack
-- Executing SetVar("SIP/300-3bb9", "FROMCONTEXT=exten-vm") in new
stack
-- Executing Macro("SIP/300-3bb9", "record-enable|204|IN") in new stack
-- Executing GotoIf("SIP/300-3bb9", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/300-3bb9",
"recordingcheck|20060228-133504|1141151704.8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060228-133504|1141151704.8: Inbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/300-3bb9", "No recording needed") in new stack
-- Executing Macro("SIP/300-3bb9", "dial|15|tr|204") in new stack
-- Executing GotoIf("SIP/300-3bb9", "0?4:2") in new stack
-- Goto (macro-dial,s,2)
-- Executing GotoIf("SIP/300-3bb9", "0?5:4") in new stack
-- Goto (macro-dial,s,4)
-- Executing AGI("SIP/300-3bb9", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: callingani2 = 0
-- dialparties.agi: accountcode =
-- dialparties.agi: channel = SIP/300-3bb9
-- dialparties.agi: callerid = 300
-- dialparties.agi: context = macro-dial
-- dialparties.agi: callington = 0
-- dialparties.agi: dnid = 204
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: calleridname = ht488
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: uniqueid = 1141151704.8
-- dialparties.agi: callingpres = 0
-- dialparties.agi: type = SIP
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: callingtns = 0
-- dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name and number are '300'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 204 to extension map
-- dialparties.agi: Extension 204 cf is disabled
-- dialparties.agi: Extension 204 do not disturb is disabled
-- dialparties.agi: Checking CW and CFB status for extension 204
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 204 is available...skipping checks
-- dialparties.agi: DbSet CALLTRACE/204 to 300
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/300-3bb9", "SIP/204|15|tr") in new stack
-- Called 204
-- SIP/204-1d0a is ringing
-- SIP/204-1d0a answered SIP/300-3bb9
-- Attempting native bridge of SIP/300-3bb9 and SIP/204-1d0a
-- Started music on hold, class 'default', on channel 'SIP/300-3bb9'
-- Stopped music on hold on SIP/300-3bb9
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/300-3bb9' in macro 'dial'
== Spawn extension (macro-exten-vm, s, 4) exited non-zero on
'SIP/300-3bb9' in macro 'exten-vm'
== Spawn extension (from-internal, 204, 1) exited non-zero on
'SIP/300-3bb9'
-- Executing Macro("SIP/300-3bb9", "hangupcall") in new stack
-- Executing ResetCDR("SIP/300-3bb9", "w") in new stack
-- Executing NoCDR("SIP/300-3bb9", "") in new stack
-- Executing Wait("SIP/300-3bb9", "5") in new stack
-- Executing Hangup("SIP/300-3bb9", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/300-3bb9' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/300-3bb9'
asterisk1*CLI>
but the call is send only to extension 204.
Is possible that the Inbound routing routed only "from-pstn"? My FXO
(300) is in a from-internal!
Where is the problem?
Thanks
--
Pasqualotto Enrico
email: pasqu at linux.it
web: http://www.pasqualotto.org
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