I have 2 FXO channels from which I want to route incoming calls to
different contexts in extensions.conf.  I edited the context entries in
dahdi-channels.conf and created matching entries in extensions.conf. 
One channel is routed to the new context as I want, but the other
channel is stuck going to the default "from-pstn" context no matter what
I do.

Can anyone see what I've missed?

>From  dahdi-channels.conf:
;;; line="3 WCTDM/4/2 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn-3
channel => 3
callerid=
group=
context=default

;;; line="4 WCTDM/4/3 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn-4
channel => 4
callerid=
group=
context=default

>From extensions.conf:
[from-pstn]
exten => s,1,Wait(1)
; exten => s,n,Answer
exten => s,n,Dial(SIP/1000)
exten => s,n,Hangup

[from-pstn-3]
exten => s,1,Wait(1)
;exten => s,n,Answer
exten => s,n,Dial(SIP/1000)
exten => s,n,Hangup

[from-pstn-4]
exten => s,1,Wait(1)
exten => s,n,Answer
exten => s,n,Dial(SIP/1000)
exten => s,n,Voicemail(1000,u)
exten => s,n,Hangup

>From the debug output:
chan_dahdi.c:     -- Starting simple switch on 'DAHDI/4-1'
pbx.c:     -- Executing [...@from-pstn-4:1] Wait("DAHDI/4-1", "1") in new
stack
pbx.c:     -- Executing [...@from-pstn-4:2] Answer("DAHDI/4-1", "") in new
stack
pbx.c:     -- Executing [...@from-pstn-4:3] Dial("DAHDI/4-1", "SIP/1000")
in new stack
netsock.c:   == Using SIP RTP TOS bits 184
netsock.c:   == Using SIP RTP CoS mark 5
app_dial.c:     -- Called 1000
app_dial.c:     -- SIP/1000-00000012 is ringing
pbx.c:   == Spawn extension (from-pstn-4, s, 3) exited non-zero on
'DAHDI/4-1'
chan_dahdi.c:     -- Hungup 'DAHDI/4-1'
chan_dahdi.c:     -- Starting simple switch on 'DAHDI/4-1'
pbx.c:     -- Executing [...@from-pstn-4:1] Wait("DAHDI/4-1", "1") in new
stack
pbx.c:     -- Executing [...@from-pstn-4:2] Answer("DAHDI/4-1", "") in new
stack
pbx.c:     -- Executing [...@from-pstn-4:3] Dial("DAHDI/4-1", "SIP/1000")
in new stack
netsock.c:   == Using SIP RTP TOS bits 184
netsock.c:   == Using SIP RTP CoS mark 5
app_dial.c:     -- Called 1000
app_dial.c:     -- SIP/1000-00000013 is ringing
pbx.c:   == Spawn extension (from-pstn-4, s, 3) exited non-zero on
'DAHDI/4-1'
chan_dahdi.c:     -- Hungup 'DAHDI/4-1'
chan_dahdi.c:     -- Starting simple switch on 'DAHDI/3-1'
pbx.c:     -- Executing [...@from-pstn:1] Wait("DAHDI/3-1", "1") in new stack
pbx.c:     -- Executing [...@from-pstn:2] Dial("DAHDI/3-1", "SIP/1000") in
new stack
netsock.c:   == Using SIP RTP TOS bits 184
netsock.c:   == Using SIP RTP CoS mark 5
app_dial.c:     -- Called 1000
app_dial.c:     -- SIP/1000-00000014 is ringing
pbx.c:   == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/3-1'
chan_dahdi.c:     -- Hungup 'DAHDI/3-1'


-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to