I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Dial(SIP/8014362345,60,r) exten => 7228888,104,Dial(SIP/8014363456,60,r) exten => 7228888,105,Dial(SIP/8014364567,60,r) exten => 7228888,106,Dial(SIP/8014365678,60,r) exten => 7228888,107,Dial(SIP/8014366789,60,r) exten => 7228888,108,Dial(SIP/8014369876,60,r) exten => 7228888,109,Dial(SIP/8014368765,60,r) exten => 7228888,110,Congestion exten => 7228888,111,Hangup OR [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,203,Dial(SIP/8014362345,60,r) exten => 7228888,304,Dial(SIP/8014363456,60,r) exten => 7228888,405,Dial(SIP/8014364567,60,r) exten => 7228888,506,Dial(SIP/8014365678,60,r) exten => 7228888,607,Dial(SIP/8014366789,60,r) exten => 7228888,708,Dial(SIP/8014369876,60,r) exten => 7228888,809,Dial(SIP/8014368765,60,r) exten => 7228888,910,Congestion exten => 7228888,1011,Hangup If extensions extension is busy or fails do you always increment by +100 or just the first time? Thanks Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040817/e8ce3bda/attachment.htm
-----Original Message----- From: Chris Modesitt [mailto:chris@octelecom.net] Sent: Tuesday, August 17, 2004 3:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hunt Groups Chris Modesitt wrote:> If extensions extension is busy or fails do you always increment by+100 or just the first time? Yes I believe that on timeout the next priority executed is the current priority plus 101. Theoretically what you are suggesting would work, but if you truly want the call to "hunt" between those extensions why not actually use an ACD (Also known as a Hunt Group). Asterisk has excelent ACD support and should be able to Provide your application with much more functionality. You can set it up via the agents.conf and queues.conf. Here are a few relavent links: * http://www.voip-info.org/wiki-Asterisk * http://www.voip-info.org/wiki-asterisk+config+queues.conf * http://www.voip-info.org/wiki-Asterisk+config+agents.conf * http://www.voip-info.org/wiki-Asterisk+agents * http://www.voip-info.org/wiki-Asterisk+call+queues * http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+co mmands (Under "Queue and ACD management") Hope this helps, Robert Jackson
On 11:15 AM 8/17/2004, Chris Modesitt wrote:>I have a question about how Asterisk Parses the Dial Plan. To create a >hunt-group which would be the appropriate dial plan: > >[CompanyABC] >exten => 7228888,1,Dial(SIP/8017228888,60,r) >exten => 7228888,102,Dial(SIP/8014361234,60,r) >exten => 7228888,203,Dial(SIP/8014362345,60,r) >exten => 7228888,304,Dial(SIP/8014363456,60,r) >exten => 7228888,405,Dial(SIP/8014364567,60,r) >exten => 7228888,506,Dial(SIP/8014365678,60,r) >exten => 7228888,607,Dial(SIP/8014366789,60,r) >exten => 7228888,708,Dial(SIP/8014369876,60,r) >exten => 7228888,809,Dial(SIP/8014368765,60,r) >exten => 7228888,910,Congestion >exten => 7228888,1011,Hangup > >If extensions extension is busy or fails do you always increment by +100 >or just the first time? >If the line you are trying is busy (in use and has an incomming-limit of 1, or the soft/hard phone reports back busy), then you increment by 101, however if the phone times out (60 seconds in your example) then the plan only increments by 1. so if you placed a call to 7228888 in your example which I kept above, the first sip phone would ring if it was available (dynamic and unregistered shows up as not available). If no one answers the call, then the dial plan would try to move to priority 2. However if the phone is busy (according to asterisk), then you would jump to priority 102. So, while it's not as technically 'clean' as using queues, the idea you have above will work as long as you add logic to handle a phone not being answered for 60 seconds. It will make for alot of entries under this exten as well. Use of Goto will allow you to limit the number of lines to about twice as many as you have above... exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,2,Goto(102) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Goto(203) ... etc.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040818/dcae1476/attachment.htm
What I would like to do is... exten => 1000,1,Dial(sip/1000)(zap/g1,97837560) exten=> 1000,2,Voicemail(u1000) Basically a follow me app that rings numerous interfaces and allows me to answer or it to time out and go to vmail. I didn't include the time out here as I am hoping someone can tell me where that needs to be. I really don't want to make the caller ring one interface and then the other. Ideally I would be able to press pound after answering so that it didn't continue to ring the other interface. Most of the apps that I saw do this are basically the same as forwarding the extension, any system can do that and I know asterisk is better than that. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060320/5249da32/attachment.htm
> What I would like to do is? > > exten => 1000,1,Dial(sip/1000)(zap/g1,97837560) > > exten=> 1000,2,Voicemail(u1000) > > Basically a follow me app that rings numerous interfaces and allows me > to answer or it to time out and go to vmail. I didn?t include the time > out here as I am hoping someone can tell me where that needs to be. I > really don?t want to make the caller ring one interface and then the > other. Ideally I would be able to press pound after answering so that > it didn?t continue to ring the other interface. Most of the apps that > I saw do this are basically the same as forwarding the extension, any > system can do that and I know asterisk is better than that. >Either put the Dial commands in sequence with a short timeout, or put multiple arguments to the dial command separated by "&" Option 1) exten => 1000,1,Dial(SIP/1000|15) exten => 1000,2,Dial(Zap/g1,97837560|15) rings each extension for 15 seconds option 2) exten => 1000,1,Dial(SIP/1000&Zip/g1,97837560) rings both extensions at once....first one to answer is the winner.