Re: [Asterisk-Users] fine-tuning asterisk questions
- William Piper <[EMAIL PROTECTED]> wrote: > By "gone forever" in 1.6... do you mean that even the "j" in the dial > plan won't work either? Will it just go to the next priority in the > event of a congested or busy signal? That is correct. All the 'j' options will go away, in favor of channel-variable result codes returned by the applications. > I assume "goto" will still work... right? Uhh... yeah. That would be silly to remove it :-) -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
On 6/5/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: This is true in Asterisk 1.2.x, as the default in the code is to enable jumping (but the default in the sample extensions.conf file is to have jumping turned off). In Asterisk 1.4 the default in the code will be to have jumping disabled, and it will need to be turned on globally (or on an application basis) to use be used. In Asterisk 1.6 it will be gone forever :-) By "gone forever" in 1.6... do you mean that even the "j" in the dial plan won't work either? Will it just go to the next priority in the event of a congested or busy signal? I assume "goto" will still work... right? bp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
- William Piper <[EMAIL PROTECTED]> wrote: > My apologies, I didn't realize I was speaking to someone else. > As far as I know the dialplan does not need to have the "j" option to > do N+101. I'm using 1.2.7.1 without the "j" option and it jumps fine. This is true in Asterisk 1.2.x, as the default in the code is to enable jumping (but the default in the sample extensions.conf file is to have jumping turned off). In Asterisk 1.4 the default in the code will be to have jumping disabled, and it will need to be turned on globally (or on an application basis) to use be used. In Asterisk 1.6 it will be gone forever :-) -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
My apologies, I didn't realize I was speaking to someone else. As far as I know the dialplan does not need to have the "j" option to do N+101. I'm using 1.2.7.1 without the "j" option and it jumps fine. I suppose that will work fine as long as you turn off call waiting on the phone itself. I provision everything from the server side so I don't have to provision the boxes. Unless you use the same SIP phones for everyone, it could be a pain to do on the phone side... and what happens if you want to give 1 person the ability to do call waiting without giving it to everyone? Even if you did the cfg files from a tftp, you'd still have to get the MAC address & provision a new cfg file what a pain! I say, depending on the size of your company, create a dialplan with a star feature to activate & deactivate call waiting. Just do a 'DBput" and dump the calleridnum value in the database, then do a "DBget" on your incoming dialplan to see if call waiting is activated for that user. That is super simple and leaves it up to the end user, or if you want... don't write the star feature & just provision it from the DB. That way it leaves it up to you and the end user still can't change it. bp On 6/5/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote: [EMAIL PROTECTED] wrote:>> Yes you are correct... by default asterisk will >send the call to priority >>> N+101... what is your point? You asked about turning off "call waiting".> In the example that I provided,>> if the amount of active calls is "1" then > it will forward to VM without>> dialing the exten. That is what you asked> for... right?>> bp>> Nope. I am a different poster just wanting to> clarify (for myself) that Asterisk would do exactly what the original poster > wanted without any special programming. I wasn't aware that there would be> any kind of notification to the station being called that there was a second> call incoming. Everything I've read so far just says that if the station > is in use, the call is routed to priority n + 101 as a busy call.Only if you use the j option in the dial command. In previous versionsit did it automatically:'j' -- Jump to n+101 if all of the channels were busy. --Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
[EMAIL PROTECTED] wrote: >> Yes you are correct... by default asterisk will >send the call to priority > >> N+101... what is your point? >> >> You asked about turning off "call waiting". > In the example that I provided, >> if the amount of active calls is "1" then > it will forward to VM without >> dialing the exten. That is what you asked > for... right? >> bp > > Nope. I am a different poster just wanting to > clarify (for myself) that Asterisk would do exactly what the original poster > wanted without any special programming. I wasn't aware that there would be > any kind of notification to the station being called that there was a second > call incoming. Everything I've read so far just says that if the station > is in use, the call is routed to priority n + 101 as a busy call. Only if you use the j option in the dial command. In previous versions it did it automatically: 'j' -- Jump to n+101 if all of the channels were busy. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
>Yes you are correct... by default asterisk will >send the call to priority > N+101... what is your point? > > You asked about turning off "call waiting". In the example that I provided, > if the amount of active calls is "1" then it will forward to VM without > dialing the exten. That is what you asked for... right? > > bp Nope. I am a different poster just wanting to clarify (for myself) that Asterisk would do exactly what the original poster wanted without any special programming. I wasn't aware that there would be any kind of notification to the station being called that there was a second call incoming. Everything I've read so far just says that if the station is in use, the call is routed to priority n + 101 as a busy call. Undrhil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
Yes you are correct... by default asterisk will send the call to priority N+101... what is your point? You asked about turning off "call waiting". In the example that I provided, if the amount of active calls is "1" then it will forward to VM without dialing the exten. That is what you asked for... right? bp On 5 Jun 2006 02:59:21 -, [EMAIL PROTECTED] < [EMAIL PROTECTED]> wrote: Hey. I was under the impressionthat Asterisk would, by default, send calls to priority n + 101 if the called station was busy. Is this not the case? Why would you have to set up somethingspecial for this to work?Undrhil___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fine-tuning asterisk questions
>For Problem #1:
> exten => _X.,1,SetGroup(${EXTEN})
> exten => _X.,2,GotoIf($[${GROUPCOUNT}
= 1]?104:3)
> exten => _X.,3,Dial,SIP/username
> exten => _X.,104,voicemail(u${EXTEN})
> exten => _X.,105,hangup
> This will limit the amount of incoming calls
to "1" and send everything else
> to the VM.
Hey. I was under the impression
that Asterisk would, by default, send calls to priority n + 101 if the called
station was busy. Is this not the case? Why would you have to set up something
special for this to work?
Undrhil
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Re: [Asterisk-Users] fine-tuning asterisk questions
For Problem #1:
exten => _X.,1,SetGroup(${EXTEN})exten => _X.,2,GotoIf($[${GROUPCOUNT} = 1]?104:3)exten => _X.,3,Dial,SIP/usernameexten => _X.,104,voicemail(u${EXTEN})exten => _X.,105,hangup
This will limit the amount of incoming calls to "1" and send everything else to the VM.
For Problem #2:
I'm not sure what you are asking. Perhaps post your dialplan for this problem & we will take a look.
bp
On 6/4/06, M.Hockings <[EMAIL PROTECTED]> wrote:
I have asterisk running more or less ok but I would like to turn offcall waiting and be selective about the incoming sip connections. This
is running asterisk 1.2.8 with a fxs and fxo card and a configured voip(sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk.Problem 1) if someone is on the phone already and another call comes in
for an already engaged extension I want it to go to voicemail directlyrather than have that distracting call-waiting beep going on.As far as I can tell I have turned off call waiting in the zaptel configfiles. What else should be set to avoid call-waiting ?
Problem 2) Incoming sip calls from my voip provider get rejected unlessI allow anyone to connect with sip. I have an incoming route set up withthe right DID that matches the DID that asterisk picks out but it still
rejects the call. Any suggestions about how to get this to work withoutallowing any sip connection?Mike
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[Asterisk-Users] fine-tuning asterisk questions
I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I want it to go to voicemail directly rather than have that distracting call-waiting beep going on. As far as I can tell I have turned off call waiting in the zaptel config files. What else should be set to avoid call-waiting ? Problem 2) Incoming sip calls from my voip provider get rejected unless I allow anyone to connect with sip. I have an incoming route set up with the right DID that matches the DID that asterisk picks out but it still rejects the call. Any suggestions about how to get this to work without allowing any sip connection? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
