Re: [Asterisk-Users] Very complicated dialplans?
Adnan Ahmed wrote: > We have 4 servers > => User1 move from server1 to server2 ,he registers on server2. > Dials an extension let's say 100 ,so all calls for User1 route on that > extension.Remember call comes from any of the 4 servers. > I implements that sort of functionality in different way but really want > that sort of dial plan is that possible or i am asking a dumb question Asterisk Realtime Architecture. http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime http://www.asteriskdocs.org/modules/news/article.php?storyid=28 -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On 8/6/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: Peter Svensson wrote:> On Sat, 6 Aug 2005, Robert Goodyear wrote:Can you educate us all on the appropriate circumstances in which to>>use 'r'?>>> Some devices (voip phones, softphones) do not generate in band progress > information when ringing. You will quickly find out if a particular> end device requires the 'r' option or not.>> You almost never want it enabled on a trunk line, only for terminal> devices. Almost nothing generates inband ringing. That has nothing to do with "r".--Eric Wieling * BTEL Consulting * 504-210-3699 x2120r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insertthis by default into all your dial statements as you are killing callprogress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically whereit is appropriate to do so. "r" makes it go the next step andadditionally generate ring tones where it is probably not appropriate to do so.That's great but i have few things to asking!We have 4 servers => User1 move from server1 to server2 ,he registers on server2. Dials an extension let's say 100 ,so all calls for User1 route on that extension.Remember call comes from any of the 4 servers. I implements that sort of functionality in different way but really want that sort of dial plan is that possible or i am asking a dumb question ___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Peter Svensson wrote: On Sat, 6 Aug 2005, Robert Goodyear wrote: Can you educate us all on the appropriate circumstances in which to use 'r'? Some devices (voip phones, softphones) do not generate in band progress information when ringing. You will quickly find out if a particular end device requires the 'r' option or not. You almost never want it enabled on a trunk line, only for terminal devices. Almost nothing generates inband ringing. That has nothing to do with "r". -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On Sat, 6 Aug 2005, Robert Goodyear wrote: > Can you educate us all on the appropriate circumstances in which to > use 'r'? Some devices (voip phones, softphones) do not generate in band progress information when ringing. You will quickly find out if a particular end device requires the 'r' option or not. You almost never want it enabled on a trunk line, only for terminal devices. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Eric Wieling aka ManxPower wrote: Robert Goodyear wrote: Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. Can you educate us all on the appropriate circumstances in which to use 'r'? When you want to override the normal call progress tones. For example, when a caller presses "0" in voicemail to be transfered to the user's cell phone, the cell phone telco may play a message to the caller WITHOUT ANSWERING the call. One common message is something like "The subscriber you have dialed is either out of the area or has their phone turned off". I don't want callers to hear that message and hangup. So in this one specific situation I use the "r" option to dial so the caller hears a ringing tone no matter what the carrier sends back. Then the Dial timeout can expire and the caller can be sent back to the user's mailbox (assuming the cell carrier didn't answer the call and send it to the cell phones voicemail). This example only works on PRI or VoIP -> PRI connections, BTW. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Robert Goodyear wrote: Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. Can you educate us all on the appropriate circumstances in which to use 'r'? When you want to override the normal call progress tones. For example, when a caller presses "0" in voicemail to be transfered to the user's cell phone, the cell phone telco may play a message to the caller WITHOUT ANSWERING the call. One common message is something like "The subscriber you have dialed is either out of the area or has their phone turned off". I don't want callers to hear that message and hangup. So in this one specific situation I use the "r" option to dial so the caller hears a ringing tone no matter what the carrier sends back. Then the Dial timeout can expire and the caller can be sent back to the user's mailbox (assuming the cell carrier didn't answer the call and send it to the cell phones voicemail). -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. Can you educate us all on the appropriate circumstances in which to use 'r'? Thx, -Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Zachary Whitley wrote: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. Ok, that response made sense. Thank you. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Thank you all for the extensive help for getting me started on the dialplan. - Arik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Zachary Whitley wrote: On Sat, 2005-08-06 at 13:02 -0400, Doug Lytle wrote: Andrew Kohlsmith wrote: On Friday 05 August 2005 21:31, Doug Lytle wrote: exten => s,1,Dial(SIP/PHONE1,15,rt) exten => s,2,Dial(SIP/PHONE4,15,rt) Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. This needs to be in the info for "show application dial" --Eric -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Only terrorists use the "r" option to Dial. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On Sat, 2005-08-06 at 13:02 -0400, Doug Lytle wrote: > Andrew Kohlsmith wrote: > > >On Friday 05 August 2005 21:31, Doug Lytle wrote: > > > > > >>exten => s,1,Dial(SIP/PHONE1,15,rt) > >>exten => s,2,Dial(SIP/PHONE4,15,rt) > >> > >> > > > >Using 'r' flags makes baby Jesus cry. Stop doing that. > > > > > > > > > > Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Andrew Kohlsmith wrote: On Friday 05 August 2005 21:31, Doug Lytle wrote: exten => s,1,Dial(SIP/PHONE1,15,rt) exten => s,2,Dial(SIP/PHONE4,15,rt) Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On Friday 05 August 2005 21:10, Arik Funke wrote: > how can I implement a dial plan like the following: Well you've scripted it out very clearly... now just use the dialplan applications and make it exactly as you've said. > 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no > answer after 15 sec also ring phones 4 and 5 I'd define some variables. Say Phones 1, 2 and 3 are on Zap channels 1, 2 and 3, and that Phone 4 and 5 are IAX phones. PHONE1=Zap/1 PHONE2=Zap/2 PHONE3=Zap/3 PHONE4=IAX2/phone4 PHONE5=IAX2/phone5 I'm assuming that the call is coming in on Zap channel 4 and you're not getting an extension # (DID) from the telco so the extension you'll hit is the 's' extension: exten => s,1,GotoIfTime(9:00-20:00|mon-fri|*|*?nine2eight,s,1) > 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if > no answer after 20 sec also ring phones 2 and 3 exten => s,2,GotoIfTime(0:00-9:00|mon-fri|*|*?midnight2nine,s,1) exten => s,3,GotoIfTime(20:00-23:59|mon-fri|*|*?eight2midnight,s,1) > 3. ring phone 1 saturday and sunday all day exten => s,4,GotoIfTime(*|sat-sun|*|*?weekends,s,1) That is the decision logic for those three rules. Now you have to define the four contexts, which I've called "nine2eight," "midnight2nine," and "eight2midnight." These contexts will dial the lines as you specify. Example: [nine2eight] exten => s,1,Dial(PHONE1&PHONE2&PHONE3,15,t) exten => s,2,Dial(PHONE1&PHONE2&PHONE3&PHONE4&PHONE5,,t) Since midnight2nine and eight2midnight contexts are the same, don't repeat your logic -- Put the actual steps in midnight2nine and make your eight2midnight context like this: [eight2midnight] exten => s,1,Goto(midnight2nine,s,1) That way if you change what you want to do for that step you don' thave to change it in two places. Of course, I think that you might be better off naming these contexts for some *function* rather than the times, but I don't know what the functions are. :-) > I do not need a in detail answer for each of the three cases... I hope I > am smart enough to generalise the answer... I just don't see an easy way > to do this right now. Hopefully this gets you started. As I said you've done an excellent job of describing what you want in small enough blocks to almost directly convert them to Asterisk dialplan logic steps. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On Friday 05 August 2005 21:31, Doug Lytle wrote: > exten => s,1,Dial(SIP/PHONE1,15,rt) > exten => s,2,Dial(SIP/PHONE4,15,rt) Using 'r' flags makes baby Jesus cry. Stop doing that. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
Arik Funke wrote: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 [incoming] exten => s,1,GotoIfTime(09:00-20:00|mon-fri|*|*?phone-rings,s,1) exten => s.2.Hangup() [phone-rings] exten => s,1,Dial(SIP/PHONE1,15,rt) exten => s,2,Dial(SIP/PHONE4,15,rt) And so on and so forth, you get the idea. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
