I have included my dial pattern - can anyone see anything that would cause this, or something in my sip.conf or h323.conf files that would override these settings?
Thanks,
Greg Oliver
[outbound]
exten => _4XXX,1,NoOp("Route to CCM1" ${EXTEN} )
exten => _4XXX,2,Dial(H323/${EXTEN})
exten => _4XXX,3,Congestionexten => _5XXX,1,NoOp("Route to CCM1" ${EXTEN} ) exten => _5XXX,2,Dial(H323/${EXTEN})
exten => _5XXX,3,Congestion
exten => _9NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN})
exten => _9NXXXXXXXXX,2,Dial(H323/${EXTEN})exten => _91NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN})
exten => _91NXXXXXXXXX,2,Dial(H323/${EXTEN})default context includes outbound, and contexts in sip.conf and h323.conf are using default. Like I say, call answered before ~5 seconds are fine, other than that it is transferred to 4607..
Howard Lowndes wrote:
On Wed, 2005-02-16 at 11:05, Greg Oliver wrote:
OK - I can successfully make calls from SIp phone through an asterisk 323 channel to a Cisco Call Manager and out a MGCP controlled gateway.
The problem is that if the call is not answered within ~5 seconds, * gives the message "no one is available to take your call" and disconnects the call. If I answer b4 the 5 seconds - everything is good.
Anywhere I need to set to get around this.
I have tried the t,T settings (even though the docs say no entry is forever) with no luck.
Read the doco on the Dial command again. It's noting to do with the Tt option, it's the parameter before that that you need to set to the timeout
Thanks,
Greg Oliver _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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