I see a couple of conflicting extensions as well as something I assume copy-paste malfunction. Please paste the CLI logs of the call.
On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito <[email protected]>wrote: > Thanks All. Here is my config: > > *On my Firewall NAT:* > > *I allowed the following ports: 4569,5004-5082, 10000-20000* > * > On Asterisk Box:* > > Here is the extensions.conf: > *[general] > static=yes > autofallthrough=yes > > [avaya-internal] > exten => s,1,Answer() > exten => s,2,background(pls-entr-num-uwish2-call) > exten => s,3,WaitExten() > exten => s,4,Dial(SIP/${EXTEN}) > exten => s,5,Dial(H323/${EXTEN}) > exten => s,6,PlayBack(vm-nobodyavail) > exten => s,7,HangUp() > > exten => 1000,1,Dial(SIP/1000) > exten => 1000,1,Answer() > > exten => 1000,2,PlayBack(vm-goodbye) > exten => 1000,3,HangUp() > > #Extension for recording > exten => 9000,1,Answer() > exten => 9000,2,Background(pm-to-record-phrase) > exten => 9000,3,Hangup() > #exten => 9000,3,Wait(2) > exten => 9000,4,Record(alt_recording%d:ulaw) > exten => 9000,5,Wait(2) > exten => 9000,6,Playback(${RECORDED_FILE}) > exten => 9000,7,Wait(2) > exten => 9000,8,Hangup > > exten => _XXXX,1,Dial(SIP/${EXTEN}@Avaya) > exten => _11XX,1,Dial(H323/${EXTEN}@Avaya) > > exten => _XXXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a) > exten => _XXXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia) > > exten => _XXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a) > exten => _XXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)* > > > > Regards, > Malvin > > > On 9/26/2011 9:56 PM, Ruben Rögels wrote: > > Am 26.09.2011 13:18, schrieb Malvin Rito: > > Hi list, > My call does not pass through on the first dial, I have to redial again > the number for the call to pass through. I'm not sure if the problem is > on my voip proovider or my asterisk server setup. Any thoughts pls? > > Regards, > Malvin > > Hi, > > could be a NAT related issue. > > Please be more specific about your setup. > > best regards, > Ruben > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
