It does not state it will dial forever. Ring forever maybe. You are posting portions of your extension.conf for outgoing calls from Asterisk only. I don't see anything here that is for incoming calls and forwarding to 4607 when the call is not answered.
Lyle ----- Original Message ----- From: "Greg Oliver" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Friday, February 18, 2005 3:58 PM Subject: Re: [Asterisk-Users] Asterisk "no one is available to take your call" > True, but it also states that with no timeout value that it will dial > until the caller hangs up. > > 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,Congestion > > exten => _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 > _______________________________________________ > 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 > _______________________________________________ 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
