Yves A.
2013-Feb-12 10:55 UTC
[asterisk-users] Asterisk Realtime Extension... strange behaviour
Hi,
I encountered a strange behaviour using realtime extensions... (on
Asterisk 11.2)
when I use the following static dialplan, everything works as expected..:
[from-sip]
exten => 110,1,Dial(DAHDI/g0/${EXTEN})
exten => 112,1,Dial(DAHDI/g0/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _X.,1,Dial(DAHDI/g0/${EXTEN})
will say... if a sip phone calls "110" or "112" the call is
routed into
PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet
to the corresponding SIP user
and if a sip phone calls any other number the call should be routed into
PSTN... thats ok and works as expected.
when I change to realtime:
[from-sip]
switch => Realtime
and put the diaplan into the database
id context exten priority app appdata
"1" "from-sip" "110" "1"
"Dial" "DAHDI/g0/${EXTEN}"
"2" "from-sip" "112" "1"
"Dial" "DAHDI/g0/${EXTEN}"
"3" "from-sip" "_XXX" "1"
"Dial" "SIP/${EXTEN}"
"4" "from-sip" "_X." "1"
"Dial" "DAHDI/g0/${EXTEN}"
only the emergency calls work and any other call goes to DAHDI... I cant
reach any other SIP phone.
Even when swapping the content of the rows 3 and 4 in the database to
id context exten priority app appdata
"1" "from-sip" "110" "1"
"Dial" "DAHDI/g0/${EXTEN}"
"2" "from-sip" "112" "1"
"Dial" "DAHDI/g0/${EXTEN}"
"3" "from-sip" "_X." "1"
"Dial" "DAHDI/g0/${EXTEN}"
"4" "from-sip" "_XXX" "1"
"Dial" "SIP/${EXTEN}"
makes no difference...
I thought, using realtime extensions would read the dialplan from top to
bottom, ordered by "id"... but it
seems to be ignored somehow and the extension "_X." catches the calls
before the extensionpattern "_XXX" is reached.
I _could_ avoid this be prefixing "external" numbers with a leading 0
for example... but I dont want to... as I said.. using
static extension via extensions.conf the dialplan works as expected...
Am I missing something?
regards,
yves
Frank
2013-Feb-12 13:41 UTC
[asterisk-users] Asterisk Realtime Extension... strange behaviour
Remove the line _X. , and try 3 digits other than 110 112 , let us know if it works. On 2/12/13 5:55 AM, Yves A. wrote:> Hi, > > I encountered a strange behaviour using realtime extensions... (on > Asterisk 11.2) > > when I use the following static dialplan, everything works as expected..: > > [from-sip] > exten => 110,1,Dial(DAHDI/g0/${EXTEN}) > exten => 112,1,Dial(DAHDI/g0/${EXTEN}) > exten => _XXX,1,Dial(SIP/${EXTEN}) > exten => _X.,1,Dial(DAHDI/g0/${EXTEN}) > > will say... if a sip phone calls "110" or "112" the call is routed into > PSTN (german emergency call) > if a sip phone calls any three digit number, the call should be routet > to the corresponding SIP user > and if a sip phone calls any other number the call should be routed into > PSTN... thats ok and works as expected. > > when I change to realtime: > [from-sip] > switch => Realtime > > and put the diaplan into the database > id context exten priority app appdata > "1" "from-sip" "110" "1" "Dial" "DAHDI/g0/${EXTEN}" > "2" "from-sip" "112" "1" "Dial" "DAHDI/g0/${EXTEN}" > "3" "from-sip" "_XXX" "1" "Dial" "SIP/${EXTEN}" > "4" "from-sip" "_X." "1" "Dial" "DAHDI/g0/${EXTEN}" > > only the emergency calls work and any other call goes to DAHDI... I cant > reach any other SIP phone. > Even when swapping the content of the rows 3 and 4 in the database to > id context exten priority app appdata > "1" "from-sip" "110" "1" "Dial" "DAHDI/g0/${EXTEN}" > "2" "from-sip" "112" "1" "Dial" "DAHDI/g0/${EXTEN}" > "3" "from-sip" "_X." "1" "Dial" "DAHDI/g0/${EXTEN}" > "4" "from-sip" "_XXX" "1" "Dial" "SIP/${EXTEN}" > > makes no difference... > I thought, using realtime extensions would read the dialplan from top to > bottom, ordered by "id"... but it > seems to be ignored somehow and the extension "_X." catches the calls > before the extensionpattern "_XXX" is reached. > > I _could_ avoid this be prefixing "external" numbers with a leading 0 > for example... but I dont want to... as I said.. using > static extension via extensions.conf the dialplan works as expected... > > Am I missing something? > > regards, > yves > > > > -- > _____________________________________________________________________ > -- 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