Dears, My scenario is to accept the call from user ?Answer the call -?play moh? dial(SIP/Trunk,XXXXX) The problem is when the user send the bye the trunk call will not hangup How to solve this issue exten => 446696,1,Ringing exten => 446696,n,Answer() exten => 446696,n,Wait(2) exten => 446696,n,Playback(Welcome) exten => 446696,n,Dial(SIP/Trunk/${EXTEN},300) exten => 446696,n,Hangup How to solve such issue Thanks in advance -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 3238 bytes Desc: not available URL: <lists.digium.com/pipermail/asterisk-users/attachments/20120608/7e0d5297/attachment.bin>
try the dial option 'g' that carries on with dialplan On 8 June 2012 09:26, Khaled W. Chehab <kchehab at xplorium.com> wrote:> Dears, > > > > My scenario is to accept the call from user ?Answer the call -?play moh? > dial(SIP/Trunk,XXXXX) > > The problem is when the user send the bye the trunk call will not hangup > > How to solve this issue > > > > > > exten => 446696,1,Ringing > > exten => 446696,n,Answer() > > exten => 446696,n,Wait(2) > > exten => 446696,n,Playback(Welcome) > > exten => 446696,n,Dial(SIP/Trunk/${EXTEN},300) > > exten => 446696,n,Hangup > > > > > > How to solve such issue > > Thanks in advance > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <lists.digium.com/pipermail/asterisk-users/attachments/20120608/2b903f86/attachment.htm>