Hi All,
While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in
my dialplan.
I had 2 sip extensions 555 and 666 and I called from 555 to 666, but
unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI.
Would you please let me know what is wrong with my dialplan and/or what else
should be done to get the value of ${RTPAUDIOQOS}?
Following is my dialplan context where my call landed....
[incoming_vpbx]
exten => _x.,1,NoOp(A call has come)
exten => _x.,n,Noop(============${RTPAUDIOQOS})
exten => _x.,n,Dial(SIP/666,30,m)
exten => _x.,n,Hangup()
exten => h,1,Noop(***************${RTPAUDIOQOS})
And here is what appeared on CLI...
-- Executing [12 at incoming_vpbx:1] NoOp("SIP/555-b7a80948",
"A call has
come") in new stack
-- Executing [12 at incoming_vpbx:2] NoOp("SIP/555-b7a80948",
"============") in new stack
-- Executing [12 at incoming_vpbx:3] Dial("SIP/555-b7a80948",
"SIP/666,30,m") in new stack
== Using SIP RTP CoS mark 5
-- Called 666
-- Started music on hold, class 'default', on SIP/555-b7a80948
-- SIP/666-089cb090 is ringing
-- SIP/666-089cb090 answered SIP/555-b7a80948
-- Stopped music on hold on SIP/555-b7a80948
-- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090
-- Executing [h at incoming_vpbx:1] NoOp("SIP/555-b7a80948",
"***************") in new stack
Thanking you...
---Asterisk User
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Do you have canreinvite=no in sip.conf? Maybe the variable is only set if Asterisk is actually relaying RTP too. regards klaus Asterisk User wrote:> Hi All, > > While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use > it in my dialplan. > I had 2 sip extensions 555 and 666 and I called from 555 to 666, but > unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI. > > Would you please let me know what is wrong with my dialplan and/or what > else should be done to get the value of ${RTPAUDIOQOS}? > > Following is my dialplan context where my call landed.... > > [incoming_vpbx] > exten => _x.,1,NoOp(A call has come) > exten => _x.,n,Noop(============${RTPAUDIOQOS}) > > exten => _x.,n,Dial(SIP/666,30,m) > exten => _x.,n,Hangup() > exten => h,1,Noop(***************${RTPAUDIOQOS}) > > > And here is what appeared on CLI... > -- Executing [12 at incoming_vpbx:1] NoOp("SIP/555-b7a80948", "A call > has come") in new stack > -- Executing [12 at incoming_vpbx:2] NoOp("SIP/555-b7a80948", > "============") in new stack > -- Executing [12 at incoming_vpbx:3] Dial("SIP/555-b7a80948", > "SIP/666,30,m") in new stack > == Using SIP RTP CoS mark 5 > -- Called 666 > -- Started music on hold, class 'default', on SIP/555-b7a80948 > -- SIP/666-089cb090 is ringing > -- SIP/666-089cb090 answered SIP/555-b7a80948 > -- Stopped music on hold on SIP/555-b7a80948 > -- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090 > -- Executing [h at incoming_vpbx:1] NoOp("SIP/555-b7a80948", > "***************") in new stack > > > Thanking you... > > ---Asterisk User > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Klaus,
Yes I do have set canreinvite=no in sip.conf.
One more thing I noticed is following two cases when I replaced exten =>
_x.,n,Dial(SIP/666,30,m) with .exten => _x.,n,Dial(SIP/666,30,me)
(1) When called extension(666) receives and hangs up the call.
-- Executing [12 at incoming_vpbx:1] NoOp("SIP/555-b7918e68", "A
call has
come") in new stack
-- Executing [12 at incoming_vpbx:2] NoOp("SIP/555-b7918e68",
"============") in new stack
-- Executing [12 at incoming_vpbx:3] Dial("SIP/555-b7918e68",
"SIP/666,30,me") in new stack
== Using SIP RTP CoS mark 5
-- Called 666
-- Started music on hold, class 'default', on SIP/555-b7918e68
-- SIP/666-09830108 is ringing
-- SIP/666-09830108 answered SIP/555-b7918e68
-- Stopped music on hold on SIP/555-b7918e68
-- Packet2Packet bridging SIP/555-b7918e68 and SIP/666-09830108
-- Executing [h at incoming_vpbx:1] NoOp("SIP/555-b7918e68",
"**************") in new stack
-- Executing [h at incoming_vpbx:1]
NoOp("[1;35;40mSIP/666-09830108",
"**************ssrc=1245221053;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000")
in new stack
== Spawn extension (incoming_vpbx, 12, 3) exited non-zero on
'SIP/555-b7918e68'
(2)When called extension(666) receives and caller extension(555) hangs up
the call.
-- Executing [12 at incoming_vpbx:1] NoOp("SIP/555-b7918e68",
"A call has
come") in new stack
-- Executing [12 at incoming_vpbx:2] NoOp("SIP/555-b7918e68",
"============") in new stack
-- Executing [12 at incoming_vpbx:3] Dial("SIP/555-b7918e68",
"SIP/666,30,me") in new stack
== Using SIP RTP CoS mark 5
-- Called 666:00*CLI>
-- Started music on hold, class 'default', on SIP/555-b7918e68
-- SIP/666-09830108 is ringing
-- SIP/666-09830108 answered SIP/555-b7918e68
-- Stopped music on hold on SIP/555-b7918e68
-- Packet2Packet bridging SIP/555-b7918e68 and SIP/666-09830108
-- Started music on hold, class 'default', on SIP/555-b7918e68
-- Stopped music on hold on SIP/555-b7918e68
-- Executing [h at incoming_vpbx:1] NoOp("SIP/555-b7918e68",
"**************ssrc=1405826681;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=101;rlp=0;rtt=0.000000")
in new stack
-- Executing [h at incoming_vpbx:1] NoOp("SIP/666-09830108",
"**************") in new stack
== Spawn extension (incoming_vpbx, 12, 3) exited non-zero on
'SIP/555-b7918e68'
So it looks like it has something to do with the way a call is hungup.
Has anybody else any idea?
Thanks,
---Asterisk User
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