Well, I¹ve tried this, and something just isn¹t right. Here¹s the context from "dialplan show," so I know it¹s loaded anyway: [ Context 'PressTwo' created by 'pbx_config' ] '*' => 1. Goto(accept|s|1) [pbx_config] '1' => 1. ForkCDR(v,s(fullcmd=${Data})) [pbx_config] 2. Background(${Data}) [pbx_config] 3. Background(repeatmsg) [pbx_config] 4. WaitExten(5,m) [pbx_config] 5. Hangup() [pbx_config] '2' => 1. Background(calllater) [pbx_config] 2. ForkCDR(v,s(reject=${Data})) [pbx_config] 3. Hangup() [pbx_config] '3' => 1. Goto(accept|1|2) [pbx_config] 'i' => 1. Goto(accept|s|1) [pbx_config] 's' => 1. Answer() [pbx_config] 2. Background(important) [pbx_config] 3. WaitExten(5,m) [pbx_config] 't' => 1. Goto(accept|s|1) [pbx_config]
I (hopefully correctly) translated your dialplan into a simple AMI command set thus: Action: Originate Channel: SIP/ShoreTel-1 Variable: "Data=/var/lib/asterisk/sounds/custom/msg1.wav" Context: PressTwo priority: 1 Number: 7979 When I hit AMI via telnet, login, and execute the above, here's the output: Response: Error Message: Originate failed Event: Newchannel Privilege: call,all Channel: SIP/ShoreTel-1-00000004 ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: from-internal Uniqueid: 1277768352.7 Event: VarSet Privilege: dialplan,all Channel: SIP/ShoreTel-1-00000004 Variable: SIPCALLID Value: 39251bb451a38e136293d70252d0d...@10.10.6.45 Uniqueid: 1277768352.7 Event: VarSet Privilege: dialplan,all Channel: SIP/ShoreTel-1-00000004 Variable: Data Value: /var/lib/asterisk/sounds/custom/msg1.wav Uniqueid: 1277768352.7 Event: NewAccountCode Privilege: call,all Channel: SIP/ShoreTel-1-00000004 Uniqueid: 1277768352.7 AccountCode: OldAccountCode: Event: NewCallerid Privilege: call,all Channel: SIP/ShoreTel-1-00000004 CallerIDNum: CallerIDName: Uniqueid: 1277768352.7 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Hangup Privilege: call,all Channel: SIP/ShoreTel-1-00000004 Uniqueid: 1277768352.7 CallerIDNum: <unknown> CallerIDName: <unknown> Cause: 17 Cause-txt: User busy Event: RTPReceiverStat Privilege: reporting,all SSRC: 0 ReceivedPackets: 0 LostPackets: 0 Jitter: 0.0000 Transit: 0.0000 RRCount: 0 Event: RTPSenderStat Privilege: reporting,all SSRC: 462309403 SentPackets: 0 LostPackets: 0 Jitter: 0 SRCount: 0 RTT: 0.000000 Event: RTPReceiverStat Privilege: reporting,all SSRC: 0 ReceivedPackets: 0 LostPackets: 0 Jitter: 0.0000 Transit: 0.0000 RRCount: 0 Event: RTPSenderStat Privilege: reporting,all SSRC: 1486119401 SentPackets: 0 LostPackets: 0 Jitter: 0 SRCount: 0 RTT: 0.000000 Thing is, I know I can dial out via that SIP trunk, and it's a test system nobody else is using, so why am I getting "User busy" here? On 6/22/10 10:31 AM, "Danny Nicholas" <da...@debsinc.com> wrote: > #1 once you¹ve got to this point, AMI would be a better option than a call > file > #2 - using AMI or a call file, you are going to want to use the context-based > method instead of application to get the most ³bang for your buck² > > I use a bigger instance of this to play a message and accept 1 or 2 from the > user > ; this context is used by AMI to play a message > [accept] > exten => s,1,Answer > exten => s,n,Background(important) > exten => s,n,WaitExten(5,m) > exten => 1,1,ForkCDR(v,s(fullcmd=${Data})) > exten => 1,n,Background(${Data}) > exten => 1,n,Background(repeatmsg) > exten => 1,n,WaitExten(5,m) > exten => 1,n,Hangup > exten => 2,1,Background(calllater) > exten => 2,n,ForkCDR(v,s(reject=${Data})) > exten => 2,n,Hangup > exten => 3,1,Goto(accept|1|2) > exten => *,1,Goto(accept|s|1) > exten => i,1,Goto(accept|s|1) > exten => t,1,Goto(accept|s|1) > > here¹s the call file > Action => 'Originate', > Channel => DAHDI/1, > Variable => "Data=/tmp/test.gsm², > Exten => 'SIP/170', > Context => 'accept', > priority => 1, > Number => 5551212 > Using the accept context, 5551212 is called on DAHDI/1 and user hears > important.gsm. then they press 1 to hear test.gsm or 2 to hear it later. > > Hope this is helpful > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Ely > Sent: Tuesday, June 22, 2010 12:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Call file structure and syntax > > Hi there, > > I¹ve been looking to do an outbound dialer for systems alerting, etc. and have > in large part followed the recipe here: > http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out > > That and the associated pages at voip-info give a basic set of recipes for > callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen, > Madsen, & Smith can I find a detailed discussion of what goes into a callfile, > how to get it to do things like interact with the shell (I¹d like ³Press 2² in > my outbound call to do something of value), etc. I¹ve googled around but > haven¹t found what I¹m looking for, just other people¹s ³Hello World² > callfiles. As of now, I can make outbound calls well enough, but want more... > > Can someone point me in the right direction for this? > > Thanks, > Mike > > -- _____________________________________________________________________ -- 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