Le 10/01/2010 07:53, Tilghman Lesher a écrit :
> On Saturday 09 January 2010 15:22:29 Benoit wrote:
>
>> I'm playing around with asterisk 1.6.2.0 and the first try was to
>> replace my now non-functionning
>> 'app-realtime' macro which emulated RealTime with REALTIME_HASH()
>>
>> There is very few documentation on the subject except for this bug report:
>> https://issues.asterisk.org/view.php?id=13651#c94998
>>
>> However when i try this syntax:
>> Set(HASH(info)=${REALTIME_HASH(call_info,exten,${dest})});
>> the syntax doesn't seem to be happy:
>>
>> -- Executing [...@appel_deb:8] Set("SIP/maverick-00000000",
>> "HASH(info)=,101,maverick,0,0,max,0,0,123456,123654") in new stack
>> [Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
>> pbx_builtin_setvar_multiple: MSet: ignoring entry '101' with no '=' (in
>> s...@appel_deb:8
>> [Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
>> pbx_builtin_setvar_multiple: MSet: ignoring entry 'maverick' with no '='
>> (in s...@appel_deb:8
>> [Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
>> pbx_builtin_setvar_multiple: MSet: ignoring entry '0' with no '=' (in
>> s...@appel_deb:8
>> [Jan 9 22:07:25] WARNING[27801]: pbx.c:9107
>> pbx_builtin_setvar_multiple: MSet: ignoring entry '0' with no '=' (in
>> s...@appel_deb:8
>> ....
>>
>> I had to do the following:
>> Set(HASH(info)="${REALTIME_HASH(call_info,exten,${dest})}"); (adding
>> of double quote)
>>
> Yes, this is because you're on a machine that you upgraded from 1.4. This
> makes Set get the old 1.4 behavior that I tried to leave behind. In your
> asterisk.conf file, create or modify the following section:
>
> [compat]
> app_set=1.6
>
> and it will start working beautifully, in an intuitively obvious way.
>
Hi,
Thank you it does indeed fix the problem, i should have read more
carefully the UPGRADE-1.6.txt before posting :(
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under asterisk 1.6.
Asterisk does receive inbound calls, with extensions informations and
all but when going to the point
of actually dialing a phone and connect it to the call it look like
"stuck", well not totally stuck since the
Dial's timeout is working and all but the sip phone isn't ringing,
asterisk isn't reporting that the phone
is ringing and the call end's up to voicemail which is working at least
for emitting audio, i have not tested
recording
Calling through the TE220 (working):
-- Executing [...@appel_deb:46] Dial("DAHDI/1-1", "SIP/benoit,8,tTwW,")
in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called benoit
-- SIP/benoit-00000032 is ringing
-- Channel 0/1, span 1 got hangup request, cause 16
Calling through the B410p (not working):
-- Executing [...@appel_deb:46] Dial("DAHDI/63-1",
"SIP/benoit,8,tTwW,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called benoit
-- Channel 0/1, span 3 got hangup request, cause 16
Any idea ?
regards
--
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