Benjamin Jacob
2008-May-28 10:31 UTC
[asterisk-users] any pointers to mute/unmute a channel
Hello ppl, Haven't yet found a way to achieve mute/unmute-ing a channel. Anybody ever attempted this? Be it even code modification or anything? A simple re-INVITE mid-session(when required) would have done, but alas, my previous questions on that one too got no responses. All help appreciated. - Benjamin Jacob.
RoLaNd RoLaNd
2008-May-28 12:45 UTC
[asterisk-users] pbx.c:2494 __ast_pbx_run: Invalid extension
Hello all,
im having trouble directing incoming calls to specific extensions after the
WAITEXTEN rule has been executed.
for example when i call in and asterisk picks up, i hear the msg.. if try to
call 105 for example, it just takers 10.. and sometimes even just "1"
please see the following error as well as my sip.conf and extensions.conf
CLI sip debugging ERROR:
-- Executing [120 at spa:1] Goto("SIP/101-b5f65a78",
"sipura-line|120|1") in new stack
-- Goto (sipura-line,120,1)
-- Executing [120 at sipura-line:1] Answer("SIP/101-b5f65a78",
"") in new stack
-- Executing [120 at sipura-line:2] Playback("SIP/101-b5f65a78",
"silence/1") in new stack
-- <SIP/101-b5f65a78> Playing 'silence/1' (language
'en')
-- Executing [120 at sipura-line:3] BackGround("SIP/101-b5f65a78",
"simzy") in new stack
-- <SIP/101-b5f65a78> Playing 'simzy' (language 'en')
-- Executing [120 at sipura-line:4] WaitExten("SIP/101-b5f65a78",
"5") in new stack
[May 28 14:41:25] WARNING[13091]: pbx.c:2494 __ast_pbx_run: Invalid extension
'10', but no rule 'i' in context 'sipura-line'
sip.conf
[100]
secret=1234
allow=all
host=dynamic
type=friend
context=sipura-line
[101]
secret=1234
allow=all
host=dynamic
type=friend
context=spa
[102]
secret=1234
allow=all
host=dynamic
type=friend
context=spa
[103]
secret=1234
allow=all
host=dynamic
type=friend
context=spa
[120]
secret=1234
allow=all
host=dynamic
type=friend
context=sipura-line
[105]
secret=1234
allow=all
host=dynamic
type=friend
context=sipura-line
extensions.conf:
[sipura-line]
exten => 120,1,Answer() ; Answer inbound calls
exten => 120,2,Playback(silence/1)
exten => 120,3,Background(simzy) ; input an extension
exten => 120,n,WaitExten(5) ; Adjust wait, default 5 sec
exten => 120,n,Goto(spa,${EXTEN}@192.168.0.111:5061,1) ; Goto the correct
extension
exten => 120,n,Hangup() ; End the call
[spa]
exten =>_120,1,GoTo(sipura-line,${EXTEN},1)
Exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _0.,1,Dial(SIP/101/${EXTEN:1})
exten => _1X.,1,Dial(SIP/${EXTEN}@192.168.0.111:5061)
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try using meetme it has a built in mute function On 5/28/08, Benjamin Jacob <ben4asterisk at yahoo.com> wrote:> > Hello ppl, > > Haven't yet found a way to achieve mute/unmute-ing a channel. > Anybody ever attempted this? Be it even code modification or anything? > > A simple re-INVITE mid-session(when required) would have done, but alas, my > previous questions on that one too got no responses. > > All help appreciated. > > - Benjamin Jacob. > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >