RE: [Asterisk-Users] making * more like a normal pbx (cisco ata-186)
On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: Ive done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I dont use a throw away digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. Sure would be nice to see that extensions.conf posted... -- Robert Withrow, [EMAIL PROTECTED], +1 978 288 8256, ESN 248 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx (ciscoata-186)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Withrow
Sent: Tuesday, June 15, 2004 12:32 PM
To: Asterisk-users
Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata-
186)
On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote:
I've done something similar at home, but made my dialplan such that I
can dial either 10 or 11 digits locally. I don't use a throw away
digit at all. Any 7, 10, or 11 digit call will be appropriately
mangled and sent out the PSTN / VoIP provider.
Sure would be nice to see that extensions.conf posted...
Here you go. I'm afraid it's rather messy, as I'm in just get the thing
working mode, but I hope it can serve as a good example. It's a modified
version of the sample extensions.conf. There are a few bugs in it, since I'm
just starting to work through it.
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=myname:mypassword; IAXtel username/password
VP=IAX2/[EMAIL PROTECTED]; Voicepulse
VON=Zap/3; Vonage adaptor is plugged into the TDM400
PSTN=Zap/4 ; Bellsouth trunk
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
;exten = s,1,SetCallerID(13212063934)
exten = s,1,SetCIDName(Internal)
exten = s,2,Dial(${ARG2},20) ; Ring for 20 secs max
exten = s,3,Voicemail(u${ARG1}) ; If unavailable, voicemail
exten = s,4,Goto(default,s,1); If they press #, start over
exten = s,102,Voicemail(b${ARG1}); If busy, send to voicemail
exten = s,103,Goto(default,s,1)
; Make an outbound call through Vonage
[macro-dialout-VON];
exten = s,1,Dial,${VON}/${ARG1}
exten = s,2,Congestion
; Make an outbound call through Voicepulse
[macro-dialout-VP];
exten = s,1,SetCallerID(My Name (321) 555-1212)
exten = s,2,Dial,${VP}/${ARG1}
exten = s,3,Congestion
;
; Outbound calls will default to Vonage, then roll over to
; Voicepulse if Vonage is either busy or can't complete the call
[macro-dialout-MULTI];
exten = s,1,SetCallerID(My Name (321) 555-1212)
exten = s,2,Dial,${VON}/${ARG1}
exten = s,3,Dial,${VP}/${ARG1}
exten = s,4,Congestion
; This macro will dial out first using Voicepulse and then
; using IAXtel if VP isn't available
[macro-dialout-VP-IAXtel]
exten = s,1,SetCallerID(My Name (321) 555-1212)
exten = s,2,Dial,${VP}/${ARG1}
exten = s,3,Dial,IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
exten = s,4,Congestion
; dials an internal extension and outside number simultaneously
; First to answer gets the call
[macro-dualextn];
exten = s,1,SetCallerID(My Name (321) 555-1212)
exten = s,2,Dial(${ARG2}${VP}/${ARG3},22)
exten = s,3,Voicemail(u${ARG1})
exten = s,4,Goto(default,s,1)
exten = s,102,Voicemail(b${ARG1})
exten = s,103,Goto(default,s,1)
;
; Inbound calls from VoicePulse
;
[voicepulse]
exten = 3212063934,1,Goto(mainmenu,s,1)
; Inbound calls from Zaptel (either Vonage or PSTN)
[inbound]
exten = s,1,Dial(SIP/510Zap/1,23)
exten = s,2,Voicemail(u500)
exten = s,3,Hangup
exten = s,102,Voicemail(b500)
exten = s,103,Hangup
; I only route inbound Voicepulse to this context
[mainmenu]
exten = s,1,Answer
exten = s,2,DigitTimeout(10) ; Set Digit Timeout to 10 seconds
exten = s,3,ResponseTimeout(20); Set Response Timeout to 20 seconds
exten = s,4,Wait(2)
exten = s,5,Background(vm-extension) ; Ask them for the extension they
want
exten = t,1,Hangup
; This is the default context for internal phones, both
; SIP and analog
[default]
include = internal-extn
; always route 7005554141 through Voicepulse
exten = 17005554141,1,Macro(dialout-VP,${EXTEN})
; IAXtel numbers (i.e. other 1700 numbers)
exten = _1700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
; toll free numbers are routed first through VP then through IAXtel
exten = _1800NXX,1,Macro(dialout-VP-IAXtel,${EXTEN})
exten = _1855NXX,1,Macro(dialout-VP-IAXtel,${EXTEN})
exten = _1866NXX,1,Macro(dialout-VP-IAXtel,${EXTEN})
exten = _1877NXX,1,Macro(dialout-VP-IAXtel,${EXTEN})
exten = _1888NXX,1,Macro(dialout-VP-IAXtel,${EXTEN})
include = outbound-dial
exten = h,1,Hangup
; rules for outbound dialing
[outbound-dial]
; block 900 numbers
exten = _1900NXX,1,Congestion
exten = _1976NXX,1,Congestion
exten = _976,1,Congestion
; Here is our local dialing plan
;
exten = _1NXXNXX,1,Macro(dialout-MULTI,${EXTEN})
exten = _407NXX,1,Macro(dialout-MULTI,1${EXTEN})
exten = _321NXX,1,Macro(dialout-MULTI,1${EXTEN})
; If I dial a 9 first, the call will go out through Voicepulse
exten = _91NXXNXX,1,Macro(dialout-VP,${EXTEN:1})
exten = _9NXXNXX,1,Macro(dialout-VP,1${EXTEN:1})
exten = _9407NXX,1,Macro(dialout-VP,1${EXTEN:1})
exten = _9321NXX,1,Macro(dialout-VP,1${EXTEN:1
Re: [Asterisk-Users] making * more like a normal pbx
On Mon, 2004-06-14 at 04:54, Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone,like most pbx's do? so dial tone , 9, dialtone, then ur local num Look at ignorepat= -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx
How about cmd DISA ? Umar --- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-06-14 at 04:54, Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone,like most pbx's do? so dial tone , 9, dialtone, then ur local num Look at ignorepat= -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx
Title: Message You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 4:54 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] making * more like a normal pbxonce u press 9 is there a way to make it so it restores dial tone, like most pbx's do?sodial tone , 9, dialtone, then ur local num--Gafachi.com - referal code hunter81instant iax termination - 2 cents a minuteAlso they have a great referal program, tell them jacob, hunter81 sent you
Re: [Asterisk-Users] making * more like a normal pbx
I searched for these things, however I don't know the proper terminology, so I come on here, people give me ideas, then i look on wiki. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you On Jun 14, 2004, at 8:59 AM, Jay Milk wrote: You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 4:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] making * more like a normal pbx once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
Re: [Asterisk-Users] making * more like a normal pbx
Title: Message I think he just wants to promote gafachi.com - Original Message - From: Jay Milk To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:59 AM Subject: RE: [Asterisk-Users] making * more like a normal pbx You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 4:54 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] making * more like a normal pbxonce u press 9 is there a way to make it so it restores dial tone, like most pbx's do?sodial tone , 9, dialtone, then ur local num--Gafachi.com - referal code hunter81instant iax termination - 2 cents a minuteAlso they have a great referal program, tell them jacob, hunter81 sent you
Re: [Asterisk-Users] making * more like a normal pbx
no, i have no affiliation with them. I just think they have great service. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 10:48 AM, Steve Totaro wrote: I think he just wants to promote gafachi.com x-tad-bigger- Original Message -/x-tad-bigger x-tad-bigger /x-tad-biggerx-tad-biggerFrom:/x-tad-biggerx-tad-bigger /x-tad-biggerx-tad-biggerJay Milk/x-tad-biggerx-tad-bigger /x-tad-bigger x-tad-biggerTo:/x-tad-biggerx-tad-bigger /x-tad-biggerx-tad-bigger[EMAIL PROTECTED]/x-tad-biggerx-tad-bigger /x-tad-bigger x-tad-biggerSent:/x-tad-biggerx-tad-bigger Monday, June 14, 2004 11:59 AM/x-tad-bigger x-tad-biggerSubject:/x-tad-biggerx-tad-bigger RE: [Asterisk-Users] making * more like a normal pbx/x-tad-bigger You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 4:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] making * more like a normal pbx once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
Re: [Asterisk-Users] making * more like a normal pbx
ignore pat 9 - Original Message - From: Jacob Hunter To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 2:16 PM Subject: Re: [Asterisk-Users] making * more like a normal pbx no, i have no affiliation with them. I just think they have great service.j hunter[EMAIL PROTECTED]On Jun 14, 2004, at 10:48 AM, Steve Totaro wrote: I think he just wants to promote gafachi.com- Original Message -From: Jay Milk To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:59 AMSubject: RE: [Asterisk-Users] making * more like a normal pbxYou really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 4:54 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] making * more like a normal pbxonce u press 9 is there a way to make it so it restores dial tone, like most pbx's do?sodial tone , 9, dialtone, then ur local num--Gafachi.com - referal code hunter81instant iax termination - 2 cents a minuteAlso they have a great referal program,tell them jacob, hunter81 sent you
RE: [Asterisk-Users] making * more like a normal pbx
Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num Google asterisk ignorepat - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx (cisco ata-186)
got it.. so one more question, this is on a cisco ata-186, (SIP) so it probably wont work. I have gone through the entire dialplan portion of the manual and can't find any function to make it function in this way. I am running firmware 2.16. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 12:26 PM, Andrew Thompson wrote: Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num Google asterisk ignorepat - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx (cisco ata-186)
That's a dirty secret of VoIP. What you want to do is handled by the VoIP device. Most VoIP devices do not support this feature. I've heard a rumor that the v3.1 SIP firmware for the ATA-186 supports this feature, but I don't currently have an ATA-186 available to me to test this. On Mon, 2004-06-14 at 14:51, Jacob Hunter wrote: got it.. so one more question, this is on a cisco ata-186, (SIP) so itprobably wont work. I have gone through the entire dialplan portionof the manual and can't find any function to make it function in thisway. I am running firmware 2.16. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 12:26 PM, Andrew Thompson wrote: Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dialtone, like most pbx's do? so dial tone , 9, dialtone, then ur local num Google asterisk ignorepat - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx (cisco ata-186)
Title: Message You're probably up the creek with the ATA186. I had one of mine unlocked from Vonage, and I think it'll go on ebay and I'll upgrade to another Sipura instead. I tested hotline functionality on sipura (pick-up phone and go right into context) successfully. It's conceivable you could change the sipura's dial-plan to include a singular "9", which takes you into the incoming context, and then you could... [incoming] 9,1,Dialtone 9,2, Of course, predialing (my phones let me dial a number on display, and once I pick up the receiver, they're dialed) would now become a problem, as establishing the connection may cause the first DTMF digit after the 9 to be missed by *, and you'd have to add a pause to all speed-dial (or pre-dial) numbers. If you're setting up asterisk for home-use, consider using "1" as the indicator for outside lines and always dialing 11 digits. That's what I have done, and it's oh-so-wife-friendly, plus I can always re-dial any incoming CID number without thinking about the area code, etc. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 2:52 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] making * more like a normal pbx (cisco ata-186)got it.. so one more question, this is on a cisco ata-186, (SIP) so it probably wont work. I have gone through the entire "dialplan" portion of the manual and can't find any function to make it function in this way. I am running firmware 2.16.j hunter[EMAIL PROTECTED]On Jun 14, 2004, at 12:26 PM, Andrew Thompson wrote: Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone,like most pbx's do? sodial tone , 9, dialtone, then ur local numGoogle asterisk ignorepat-Andrew Thompsonhttp://aktzero.com/ ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] making * more like a normal pbx (cisco ata-186)
Title: Message Ive done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I dont use a throw away digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Monday, June 14, 2004 5:59 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] making * more like a normal pbx (cisco ata-186) You're probably up the creek with the ATA186. I had one of mine unlocked from Vonage, and I think it'll go on ebay and I'll upgrade to another Sipura instead. I tested hotline functionality on sipura (pick-up phone and go right into context) successfully. It's conceivable you could change the sipura's dial-plan to include a singular 9, which takes you into the incoming context, and then you could... [incoming] 9,1,Dialtone 9,2, Of course, predialing (my phones let me dial a number on display, and once I pick up the receiver, they're dialed) would now become a problem, as establishing the connection may cause the first DTMF digit after the 9 to be missed by *, and you'd have to add a pause to all speed-dial (or pre-dial) numbers. If you're setting up asterisk for home-use, consider using 1 as the indicator for outside lines and always dialing 11 digits. That's what I have done, and it's oh-so-wife-friendly, plus I can always re-dial any incoming CID number without thinking about the area code, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] making * more like a normal pbx (cisco ata-186) got it.. so one more question, this is on a cisco ata-186, (SIP) so it probably wont work. I have gone through the entire dialplan portion of the manual and can't find any function to make it function in this way. I am running firmware 2.16. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 12:26 PM, Andrew Thompson wrote: Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num Google asterisk ignorepat - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] making * more like a normal pbx (cisco ata-186)
At 3:58 PM -0500 on 6/14/04, Eric Wieling wrote: On Mon, 2004-06-14 at 14:51, Jacob Hunter wrote: got it.. so one more question, this is on a cisco ata-186, (SIP) so itprobably wont work. I have gone through the entire dialplan portionof the manual and can't find any function to make it function in thisway. I am running firmware 2.16. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 12:26 PM, Andrew Thompson wrote: Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dialtone, like most pbx's do? so dial tone , 9, dialtone, then ur local num Google asterisk ignorepat - Andrew Thompson http://aktzero.com/ That's a dirty secret of VoIP. What you want to do is handled by the VoIP device. Most VoIP devices do not support this feature. I've heard a rumor that the v3.1 SIP firmware for the ATA-186 supports this feature, but I don't currently have an ATA-186 available to me to test this. Unless you do some really nasty work with PLARs on your phones (hotline dialing) and DISA on Asterisk, and some fairly trivial dialplan construction. Doing this of course totally short-circuits the concepts of session-based media services such as IAX and SIP. However, business rules or customer requirements sometimes trump clean implementations of technology. This method will answer Jacob's original question quite nicely, though an investigation of how to configure ATA-186 devices to do PLAR, and how to use DISA, will be required on his part. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
