Re: [asterisk-users] Dial Plan Issue
One follow-up. At the end of the call, after it dis-connects I get the following error: [2015-02-10 15:33:42] NOTICE[4524]: pbx_spool.c:402 attempt_thread: Call completed to SIP/SMtrunk1/xx Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Tuesday, February 10, 2015 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dial Plan Issue I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail. Here is the printout of the log file for both boxes. Free PBX: [2015-02-10 12:12:34] VERBOSE[10502] pbx.c: -- Executing [XX@subMachine:4] Playback(SIP/trunk503out-9728, temp/0250002) in new stack [2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XX @subMachine:5] Wait(SIP/trunk503out-9728, 1) in new stack [2015-02-10 12:13:30] VERBOSE[10502] pbx.c: -- Executing [XX @subMachine:6] Playback(SIP/trunk503out-9728, temp/0250002) in new stack Standard Asterisk Build: [2015-02-10 15:01:12] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:1] SendDTMF(SIP/SMtrunk1-000f, w1w) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:2] Set(SIP/SMtrunk1-000f, IVR_MSG=temp/0250002) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx@subMachine:3] System(SIP/SMtrunk1-000f, /bin/echo -e xx,RUN2,i9,02102015145822,MACHINE,SIP/SMtrunk1-000f,02.10.2015 15.01log/outbound.txt) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] pbx.c: -- Executing [xx @subMachine:4] Playback(SIP/SMtrunk1-000f, temp/0250002) in new stack [2015-02-10 15:01:16] VERBOSE[32567][C-000f] file.c: -- SIP/SMtrunk1-000f Playing 'temp/0250002.slin' (language 'en') [2015-02-10 15:01:50] VERBOSE[32567][C-000f] pbx.c: == Spawn extension (subMachine, xx, 4) exited non-zero on 'SIP/SMtrunk1-000f' I copied the context from the FreePbx box over to the new box so the code should be the same. Any help would be appreciated. Thanks, Scott If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messa...@edwardjones.commailto:messa...@edwardjones.com along with the email address you wish to unsubscribe. For important additional information related to this email, visit www.edwardjones.com/US_email_disclosurehttp://www.edwardjones.com/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. -- _ -- 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
Re: [asterisk-users] Dial plan order of operations
On Mon, Sep 24, 2012 at 02:17:29PM -0700, Steve Edwards wrote:
On Mon, 24 Sep 2012, Asterisk Newb wrote:
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
Two suggestions:
1) Using the 'n' priority will make your dialplans more maintainable.
Asterisk 1.2 does not have it, IIRC.
1.1) Upgrade to a newer version of Asterisk :-(
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Re: [asterisk-users] Dial plan order of operations
On Monday 24 September 2012, Asterisk Newb wrote:
Hello all,
I inherited an Asterisk 1.2 machine and I have a question about the order
of operations.
I want to give people the ability to dial specifics and block others. For
example, lets say NYC
[allowed]
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
exten = _1212.,s,Goto(denied,s,1)
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
What I would like to do it allow a specific and deny the rest. Mind you the
allowed will be everything EXCEPT what is allowed. My question is, will the
above work? Please don't comment on upgrading, this is an inherited system
which I cannot update.
Asterisk always tests against the most specific (= hardest-to-match)
wildcarded extensions first, regardless of the actual order in the dialplan.
Since _1212555. is harder to match than _1212., the former will be tested
first.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Dial plan order of operations
On Mon, Sep 24, 2012 at 12:43 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
Asterisk always tests against the most specific (= hardest-to-match)
wildcarded extensions first, regardless of the actual order in the
dialplan.
Since _1212555. is harder to match than _1212., the former will be tested
first.
--
AJS
So then the proper way would be
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
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Re: [asterisk-users] Dial plan order of operations
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
My final question to the list (hopefully) will be, why doesn't this work as
documented (voip-wiki, etc):
My dialplan (Asterisk 1.2 line)
exten = _212555.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = 102,1,hangup
When I try calling, I am prompted for the pin, I enter it 3 time and rather
than it go to n+101 it allows the call through.
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Re: [asterisk-users] Dial plan order of operations
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Newb
Sent: Monday, September 24, 2012 1:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan order of operations
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
My final question to the list (hopefully) will be, why doesn't this work as
documented (voip-wiki, etc):
My dialplan (Asterisk 1.2 line)
exten = _212555.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = 102,1,hangup
When I try calling, I am prompted for the pin, I enter it 3 time and rather
than it go to n+101 it allows the call through.
Not to be harsh, but the wiki information is buyer beware unless it is by
one of the developers or a select group of contributors. Lots of things in
the wikis work as written, but may fail if the wrong tweak is applied.
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Re: [asterisk-users] Dial plan order of operations
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it.. Read the
docs for Authenticate and see what diaplan variables you can check.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Newb
Sent: Monday, September 24, 2012 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan order of operations
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
exten = s,2,Hangup
?
My final question to the list (hopefully) will be, why doesn't this work as
documented (voip-wiki, etc):
My dialplan (Asterisk 1.2 line)
exten = _212555.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/IntlPins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = 102,1,hangup
When I try calling, I am prompted for the pin, I enter it 3 time and rather
than it go to n+101 it allows the call through.
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Re: [asterisk-users] Dial plan order of operations
On Mon, Sep 24, 2012 at 2:55 PM, Eric Wieling ewiel...@nyigc.com wrote:
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it..
Read the docs for Authenticate and see what diaplan variables you can
check.
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212444.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212444.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _212333.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212333.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
exten = _X.,102,hangup
; exten = 102,1,hangup --- my screw up
So when someone dials a number to a dest (212555{444{333}}) they're asked
for a PIN 3x, if it fails now it hangs up
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Re: [asterisk-users] Dial plan order of operations
On Mon, 24 Sep 2012, Asterisk Newb wrote:
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
Two suggestions:
1) Using the 'n' priority will make your dialplans more maintainable.
2) Using a more 'explicit' pattern like '_212555' will result in a
more responsive dialplan* because Asterisk 'knows' it is looking for a 10
digit number and won't have to 'wait' to see if it has the whole number.
(For circuits that don't deliver the DID/DNIS all at once.)
Using an 'open ended' pattern could also expose you to unexpected
outcomes.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Apr 11, 2012, at 5:40 PM, list...@gmail.com wrote: And your examples should work for 1.8.10 correct? I just typed those out really quick, so there may be some syntax errors, but generally yes they should all work with 1.8.x. -- Thanks, Warren Selby, dCAP -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
Ok thanks. It seems when I add the /number it quits working and I didn’t know if it would be a 1.8 issue. I will give it another try. Thanks again. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, April 12, 2012 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID On Apr 11, 2012, at 5:40 PM, list...@gmail.com wrote: And your examples should work for 1.8.10 correct? I just typed those out really quick, so there may be some syntax errors, but generally yes they should all work with 1.8.x. -- Thanks, Warren Selby, dCAP -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
Here is an example. Let's say that I want to send all calls to a context that would answer the call via voicemail. Let's say that I want to only right a SIP phone if calls cam from a particular Area Code (maybe the Area Codes in your state). Let's say that I would want to send calls from a particular A/C and certain NNX's to a particular sales group. Does that help define the purpose of directing calls *from* different Area Codes and NNX's? I wanted to come up with total control to do with an incoming call depended upon *where* that call came from whether it be a whole A/C, A/C with particular NNX's or even down to a particular A/C/NNX/number. Hope that clarifies what I was looking help with. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston Sent: Thursday, April 05, 2012 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I have never used a wildcard match like you are attempting to do with the 702 prefix but according to voip-info.org it should work -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
Although I'm a huge fan of ex-girlfriend logic, it's probably overkill for this application. Here is what I would suggest: exten = _702.n,Goto(context1,s,1) exten = _614.,n,Goto(context2,s,1) exten = _614555.,n,Goto(context3,s,1) exten = _.,n,Dial(SIP/,25) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of list...@gmail.com Sent: Wednesday, April 11, 2012 4:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID Here is an example. Let's say that I want to send all calls to a context that would answer the call via voicemail. Let's say that I want to only right a SIP phone if calls cam from a particular Area Code (maybe the Area Codes in your state). Let's say that I would want to send calls from a particular A/C and certain NNX's to a particular sales group. Does that help define the purpose of directing calls *from* different Area Codes and NNX's? I wanted to come up with total control to do with an incoming call depended upon *where* that call came from whether it be a whole A/C, A/C with particular NNX's or even down to a particular A/C/NNX/number. Hope that clarifies what I was looking help with. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston Sent: Thursday, April 05, 2012 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I have never used a wildcard match like you are attempting to do with the 702 prefix but according to voip-info.org it should work -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
Thank you, Chad. I will check out that document now. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace Sent: Thursday, April 05, 2012 2:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID On Thu, 5 Apr 2012 13:35:51 -0400 list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) I think the n's should all be 1's, like so: exten = 614000/_702XXX,1,Goto(context1,s,1) exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000/614997,1,Dial(SIP/,25) The 'n' priority is used for subsequent lines (after the first) in the same extension, but the first one for each extension should be 1. I have seen examples where I could use a pattern like (not specifying a Caller ID info, and that works fine): exten = _X!,n,Goto(context1,s,1) exten = _X!,n,Goto(context2,s,1) exten = _X!,n,Goto(context3,s,1) I suggest you don't use _X! or _X. as a pattern, until you fully understand the security risks. In the asterisk-1.8 tarball, there's a file called README-SERIOUSLY.bestpractices.txt that explains it all. You should read that before you do anything. I am confused on how to use patterns. I would like to learn how I can take either DID and route the calls to various contexts via the CallerID (which couild be the entire DID number, an NPA only or an NPANXX. You have an example for NPA only in the line that handles area code 702. Similar for NPANXX: exten = 614000/_614999,1,Goto(context,s,1) This is all covered quite well on the voip-info wiki: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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 Thank you, Chad. I will check out that document now. -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, April 11, 2012 5:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID Although I'm a huge fan of ex-girlfriend logic, it's probably overkill for this application. Here is what I would suggest: exten = _702.n,Goto(context1,s,1) exten = _614.,n,Goto(context2,s,1) exten = _614555.,n,Goto(context3,s,1) exten = _.,n,Dial(SIP/,25) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of list...@gmail.com Sent: Wednesday, April 11, 2012 4:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID Here is an example. Let's say that I want to send all calls to a context that would answer the call via voicemail. Let's say that I want to only right a SIP phone if calls cam from a particular Area Code (maybe the Area Codes in your state). Let's say that I would want to send calls from a particular A/C and certain NNX's to a particular sales group. Does that help define the purpose of directing calls *from* different Area Codes and NNX's? I wanted to come up with total control to do with an incoming call depended upon *where* that call came from whether it be a whole A/C, A/C with particular NNX's or even down to a particular A/C/NNX/number. Hope that clarifies what I was looking help with. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston Sent: Thursday, April 05, 2012 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I have never used a wildcard match like you are attempting to do with the 702 prefix but according to voip-info.org it should work -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein Thanks, Danny. I will give that a try. -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Wed, Apr 11, 2012 at 4:11 PM, list...@gmail.com wrote:
Here is an example.
** **
Let’s say that I want to send all calls to a context that would answer the
call via voicemail.
Let’s say that I want to only right a SIP phone if calls cam from a
particular Area Code (maybe the Area Codes in your state).
Let’s say that I would want to send calls from a particular A/C and
certain NNX’s to a particular sales group.
** **
Does that help define the purpose of directing calls **from** different
Area Codes and NNX’s?
You've got a few ways you can do this:
1 - In the dialplan with ex-girlfriend logic. You should be able to use
patterns with your ex-girlfriend logic matches, as so:
exten = 15558675309/_255NXX,1,Verbose(Calls to 867-5309 from area code
255 end up here)
exten = 15558675309/_256123,1,Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
2 - In the dialplan with GotoIf logic:
exten = 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
exten = 15558675309,n,GotoIf($[${CALLERID(num):1:3}=255]?areacode255)
exten = 15558675309,n,GotoIf($[${CALLERID(num):1:6}=256123]?num256123)
exten = 15558675309,n(areacode255),Verbose(Calls to 867-5309 from area
code 255 end up here)
exten = 15558675309,n(num256123),Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
3 - Outside the dialplan with an AGI that allows you many more conditional
logic choices (plus keeps your dialplan nice and clean):
exten = 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
same = n,AGI(route_by_clid)
In your AGI, you'll be most interested in the agi_callerid environment
variable and you can control where the call goes next using the SET CONTEXT
and SET EXTENSION agi commands, or simply EXEC a GoTo command (either way
works).
Ultimately, I would go with the AGI option, because that then allows you to
do things like use a database to store your routing information, use case
statements, create routing loops, etc. It's up to you though.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
Hi Warren,
Thanks for the help. I agree I like the AGI best. I am not a programmer
and don't know the AGI piece but I would like to learn. I will try your
first two examples and then attempt to do something with AGI. I really like
the idea of using a database and keeping the dialplan as clean as possible.
Thanks very much.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Wednesday, April 11, 2012 5:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Wed, Apr 11, 2012 at 4:11 PM, list...@gmail.com wrote:
Here is an example.
Let's say that I want to send all calls to a context that would answer the
call via voicemail.
Let's say that I want to only right a SIP phone if calls cam from a
particular Area Code (maybe the Area Codes in your state).
Let's say that I would want to send calls from a particular A/C and certain
NNX's to a particular sales group.
Does that help define the purpose of directing calls *from* different Area
Codes and NNX's?
You've got a few ways you can do this:
1 - In the dialplan with ex-girlfriend logic. You should be able to use
patterns with your ex-girlfriend logic matches, as so:
exten = 15558675309/_255NXX,1,Verbose(Calls to 867-5309 from area code
255 end up here)
exten = 15558675309/_256123,1,Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
2 - In the dialplan with GotoIf logic:
exten = 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
exten = 15558675309,n,GotoIf($[${CALLERID(num):1:3}=255]?areacode255)
exten = 15558675309,n,GotoIf($[${CALLERID(num):1:6}=256123]?num256123)
exten = 15558675309,n(areacode255),Verbose(Calls to 867-5309 from area code
255 end up here)
exten = 15558675309,n(num256123),Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
3 - Outside the dialplan with an AGI that allows you many more conditional
logic choices (plus keeps your dialplan nice and clean):
exten = 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
same = n,AGI(route_by_clid)
In your AGI, you'll be most interested in the agi_callerid environment
variable and you can control where the call goes next using the SET CONTEXT
and SET EXTENSION agi commands, or simply EXEC a GoTo command (either way
works).
Ultimately, I would go with the AGI option, because that then allows you to
do things like use a database to store your routing information, use case
statements, create routing loops, etc. It's up to you though.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
--
_
-- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
And your examples should work for 1.8.10 correct?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Wednesday, April 11, 2012 5:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Wed, Apr 11, 2012 at 4:11 PM, list...@gmail.com wrote:
Here is an example.
Let's say that I want to send all calls to a context that would answer the
call via voicemail.
Let's say that I want to only right a SIP phone if calls cam from a
particular Area Code (maybe the Area Codes in your state).
Let's say that I would want to send calls from a particular A/C and certain
NNX's to a particular sales group.
Does that help define the purpose of directing calls *from* different Area
Codes and NNX's?
You've got a few ways you can do this:
1 - In the dialplan with ex-girlfriend logic. You should be able to use
patterns with your ex-girlfriend logic matches, as so:
exten = 15558675309/_255NXX,1,Verbose(Calls to 867-5309 from area code
255 end up here)
exten = 15558675309/_256123,1,Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
2 - In the dialplan with GotoIf logic:
exten = 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
exten = 15558675309,n,GotoIf($[${CALLERID(num):1:3}=255]?areacode255)
exten = 15558675309,n,GotoIf($[${CALLERID(num):1:6}=256123]?num256123)
exten = 15558675309,n(areacode255),Verbose(Calls to 867-5309 from area code
255 end up here)
exten = 15558675309,n(num256123),Verbose(Calls to 867-5309 from phone
numbers 256123 end up here)
etc.
3 - Outside the dialplan with an AGI that allows you many more conditional
logic choices (plus keeps your dialplan nice and clean):
exten = 15558675309,1,Verbose(Call from ${CALLERID(num)} to 867-5309)
same = n,AGI(route_by_clid)
In your AGI, you'll be most interested in the agi_callerid environment
variable and you can control where the call goes next using the SET CONTEXT
and SET EXTENSION agi commands, or simply EXEC a GoTo command (either way
works).
Ultimately, I would go with the AGI option, because that then allows you to
do things like use a database to store your routing information, use case
statements, create routing loops, etc. It's up to you though.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
--
_
-- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I have never used a wildcard match like you are attempting to do with the 702 prefix but according to voip-info.org it should work -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 10:52 AM, John Kiniston johnkinis...@gmail.comwrote: On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I agree, priorities are very tricky in this case, and I've spent a lot of time figuring out similar scenarios. Also you may need to use an 's' priority in some cases, where there are two potential matches that are the same priority. I'm sorry I can't think of where I have a useful code snippet for your exact case. I'd recommend starting with a fully working explicit statement then work back from there to less explicit: exten = 16027170050/8774613644,1,Goto(monkeys,s,1) Then: exten = 1602717005X/8774613644,1,Goto(monkeys,s,1) Etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
Priorities are not complicated. Each extension starts with priority 1, all additional priorities are n and you ALWAYS end your extension with a Goto or a Hangup so the call doesn't fall off your intended extension and hump into the middle of some other extension and screw everything up. You are pretty close, I think it's your priorities that are the problem. When I use Ex Girl Friend Logic I write my extensions this way: exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000,1,Dial(SIP/,25) I agree, priorities are very tricky in this case, and I've spent a lot of time figuring out similar scenarios. Also you may need to use an 's' priority in some cases, where there are two potential matches that are the same priority. I'm sorry I can't think of where I have a useful code snippet for your exact case. I'd recommend starting with a fully working explicit statement then work back from there to less explicit: exten = 16027170050/8774613644,1,Goto(monkeys,s,1) Then: exten = 1602717005X/8774613644,1,Goto(monkeys,s,1) Etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, 5 Apr 2012 13:35:51 -0400 list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten = 614000/614998,n,Goto(context3,s,1) exten = 614000/614997,n,Dial(SIP/,25) I think the n's should all be 1's, like so: exten = 614000/_702XXX,1,Goto(context1,s,1) exten = 614000/614999,1,Goto(context2,s,1) exten = 614000/614998,1,Goto(context3,s,1) exten = 614000/614997,1,Dial(SIP/,25) The 'n' priority is used for subsequent lines (after the first) in the same extension, but the first one for each extension should be 1. I have seen examples where I could use a pattern like (not specifying a Caller ID info, and that works fine): exten = _X!,n,Goto(context1,s,1) exten = _X!,n,Goto(context2,s,1) exten = _X!,n,Goto(context3,s,1) I suggest you don't use _X! or _X. as a pattern, until you fully understand the security risks. In the asterisk-1.8 tarball, there's a file called README-SERIOUSLY.bestpractices.txt that explains it all. You should read that before you do anything. I am confused on how to use patterns. I would like to learn how I can take either DID and route the calls to various contexts via the CallerID (which couild be the entire DID number, an NPA only or an NPANXX. You have an example for NPA only in the line that handles area code 702. Similar for NPANXX: exten = 614000/_614999,1,Goto(context,s,1) This is all covered quite well on the voip-info wiki: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 11:00 AM, Eric Wieling ewiel...@nyigc.com wrote: Priorities are not complicated. Each extension starts with priority 1, all additional priorities are n and you ALWAYS end your extension with a This isn't correct, there are many cases where you must use an 's' priority. Our system simply couldn't function without it. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Apr 5, 2012, at 1:23 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Apr 5, 2012 at 11:00 AM, Eric Wieling ewiel...@nyigc.com wrote: Priorities are not complicated. Each extension starts with priority 1, all additional priorities are n and you ALWAYS end your extension with a This isn't correct, there are many cases where you must use an 's' priority. Our system simply couldn't function without it. You're think of EXTENSION 's', not PRORITY. Priority is the order the call goes through the dial plan. Extension is the part of the dial plan you're traversing. Priority will always be either a number or an 'n'. exten = EXTENSION,PRIORITY,COMMAND -- Thanks, Warren Selby, dCAP-- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 12:11 PM, Eric Wieling ewiel...@nyigc.com wrote:
Are you sure you are not referring to the s extension?
Absolutely. Every time I discuss 's' priority on this list or the Asterisk
IRC channel people tell me it either doesn't exist or is wrong, but it's a
powerful under-utilized feature. It's at the core of initially routing
calls on our system.
Show an example of needing s as a priority.
This is from our system, the asterisks have been used to obscure for
privacy, they are numbers.
exten = 1602889,n,Goto(starnetworks#main|s|1)
exten = 1602400,s,Goto(starnetworks#extensions,9520,1)
exten = 1480241,s,Goto(starnetworks#extensions,9766,1)
exten = _X.,s,Goto(starnetworks#extensions|${EXTEN:7}|1)
--
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 12:13 PM, Warren Selby wcse...@selbytech.com wrote: You're think of EXTENSION 's', not PRORITY. Priority is the order the call goes through the dial plan. Extension is the part of the dial plan you're traversing. Priority will always be either a number or an 'n'. exten = EXTENSION,PRIORITY,COMMAND Nope, it's 's' priority. See my subsequent message about it. Priority CAN be an 's' and in certain situations MUST be an 's' to function. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Apr 5, 2012, at 2:32 PM, Carlos Alvarez car...@televolve.com wrote:
On Thu, Apr 5, 2012 at 12:11 PM, Eric Wieling ewiel...@nyigc.com wrote:
Are you sure you are not referring to the s extension?
Absolutely. Every time I discuss 's' priority on this list or the Asterisk
IRC channel people tell me it either doesn't exist or is wrong, but it's a
powerful under-utilized feature. It's at the core of initially routing calls
on our system.
Show an example of needing s as a priority.
This is from our system, the asterisks have been used to obscure for privacy,
they are numbers.
exten = 1602889,n,Goto(starnetworks#main|s|1)
exten = 1602400,s,Goto(starnetworks#extensions,9520,1)
exten = 1480241,s,Goto(starnetworks#extensions,9766,1)
exten = _X.,s,Goto(starnetworks#extensions|${EXTEN:7}|1)
I still don't understand what you would need this for. What version of asterisk
are you using? From voip-info.org, it says the s priority is used when
different patterns may match at the same point in the extension and act
differently for them, but couldn't you basically do the same thing with
priority labels? How would you ever end up with different patterns matching at
the same point in an extension? Where is your priority 1?
--
Thanks,
Warren Selby, dCAP--
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Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 12:57 PM, Warren Selby wcse...@selbytech.com wrote: I still don't understand what you would need this for. What version of asterisk are you using? From voip-info.org, it says the s priority is used when different patterns may match at the same point in the extension and act differently for them, but couldn't you basically do the same thing with priority labels? How would you ever end up with different patterns matching at the same point in an extension? Where is your priority 1? Well, now we get into a lot of design philosophy discussion that I really don't have time for today. I will note that Kevin Fleming wrote the 's' feature into the code before he worked at Digium and still owned the company I now run... I didn't understand his design at all for a long time, but now it's second nature. The 1 priority is in another context that pre-processes the calls, then each customer has a series of 's' priority lines for their individual DID numbers. I use the 's' priority in some DNIS/CID-based call blocking here and there. As far as versions, 1.2, 1.4, and 1.6. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
Re: [asterisk-users] dial plan with hangup cause 34
This is a FreePBX question as the Asterisk dialplan is managed by it.
I suggest to use 'extensions_override_freepbx.conf' (details in
extensions.conf) and place there your modified [macro-dialout-trunk].
HTH,
Ioan
On Fri, Feb 10, 2012 at 1:13 PM, ing.Achim Alexandru
alexandru.achi...@gmail.com wrote:
Dear Asterisk Users,
I have a question. I use asterisk 1.6 withh freepbx on ubuntu ,
compiled manually.
I want to change the route congestion message ( all-circuit-bussy)
wiyh a hangup cause 34 ( something like that in dialplan
s,n,GotoIf($[${HANGUPCAUSE} = 34]?failover,1). Have any ideas?
Thanks
Alexandru Achim
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Re: [asterisk-users] dial plan and sip
On Sat, 13 Nov 2010 20:38:30 -0500 Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? [...] exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Dial(SIP/jazzey/1703111,120,A(demo-thanks)) -- _ -- 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
Re: [asterisk-users] dial plan and sip
thank you i will try it. On Mon, Nov 15, 2010 at 4:52 PM, Chad Wallace cwall...@lodgingcompany.com wrote: On Sat, 13 Nov 2010 20:38:30 -0500 Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? [...] exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Dial(SIP/jazzey/1703111,120,A(demo-thanks)) -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
Why do you have A,(demo-thanks) shouldn't it it be A(demo-thanks)? eg: exten = s,n,Dial(SIP/jazzey/1703111,120,A(demo-thanks)) On Sat, Nov 13, 2010 at 6:38 PM, Thomas Perron thomas.per...@gmail.comwrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com908366554%3a396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com908366554%3a396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
You get into asterisk by saying asterisk -r. You then up the verbosity by saying core set verbose 3 or some such number. You the call your number and you should see the steps of your dialplan execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] dial plan and sip
Jim, Thanks. But, no joy. I set to 3, then 5. I don't think I am getting registered somewhere. The console shows nothing. The call to the DID drops after 5 seconds or so. It does not ring. I know. Basic stuff. I really think the version of this code is not robust enough. My sip.conf and extensions.conf is very simple. On Sat, Nov 13, 2010 at 10:13 PM, Jim Dickenson dicken...@cfmc.com wrote: You get into asterisk by saying asterisk -r. You then up the verbosity by saying core set verbose 3 or some such number. You the call your number and you should see the steps of your dialplan execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon Sent: Wednesday, October 27, 2010 4:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Conf Jigar, You should use Read() instead of Background() component. See attached Visual Dialplan file. Nile Finally got VDP to show me this dialplan. A Gotoif will satisfy rest of OP's request. -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
On Wed, 27 Oct 2010, Nile Kaledon wrote: You should use Read() instead of Background() component. We conf file weenies call them applications. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, October 27, 2010 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial Plan Conf On Wed, 27 Oct 2010, Nile Kaledon wrote: You should use Read() instead of Background() component. We conf file weenies call them applications. Is that like a Perl Weenie? -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
On Wed, 27 Oct 2010, Nile Kaledon wrote: You should use Read() instead of Background() component. On Wed, 27 Oct 2010, Steve Edwards wrote: We conf file weenies call them applications. On Wed, 27 Oct 2010, Danny Nicholas wrote: Is that like a Perl Weenie? Yes, and you can proudly wear as many self-congratulatory labels as you wish simultaneously. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
Hi I want that all of my call should be asked for a code . And then all call should go to a fixed extension. My application will be running there that will differentiate stream of calls. like person A enters 1234 person B enters 2345 both call will be directed to extension say 101, and from there my app will create two audio stream one is by reading code entered by caller . I am currently reading book as instructed. But it would be more helpful if you have already parsed that vdp. On Tue, Oct 26, 2010 at 2:23 AM, Nile Kaledon nile.kale...@gmail.comwrote: Hi, I just downloaded your vdp file and it's working fine on my installation (Asterisk 1.4). Can you be more specific on the issue you experienced? Nile -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial plan help
Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com Nile -- _ -- 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
Re: [asterisk-users] Dial plan help
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon Sent: Monday, October 25, 2010 12:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial plan help Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com Nile I'll have to agree that VDP is a nice tool, but it is just that - a tool. If you don't know how the dialplan and commands work, it will eventually dig you into a hole you won't get out of. -- _ -- 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
Re: [asterisk-users] Dial plan help
Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nile Kaledon *Sent:* Monday, October 25, 2010 12:06 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Dial plan help Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com Nile I’ll have to agree that VDP is a nice tool, but it is just that – a tool. If you don’t know how the dialplan and commands work, it will eventually dig you into a hole you won’t get out of. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial plan help
Chapters 4, 5 and 6 is a good start. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:aster... -- _ -- Bandwidth and Colo... -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial plan help
Un-top-posting... On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, 25 Oct 2010, Zeeshan Zakaria wrote: Chapters 4, 5 and 6 is a good start. Yep. That's where I'd start if I didn't even know enough to ask questions using the correct terminology. I always skip the first 3 chapters in any technical book because I figure the authors put them in just to fill out their commitment to the publisher so he can charge more for the book -- even when the book is available for free. I figure, why learn the foundation of a new technology when there are always mailing lists manned by volunteers waiting at my beck and call -- my time is worth more than theirs. The one thing I can't figure out is why everybody keeps adding me to their MUA kill lists... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] Dial plan help
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, October 25, 2010 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial plan help Un-top-posting... On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote: Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, 25 Oct 2010, Zeeshan Zakaria wrote: Chapters 4, 5 and 6 is a good start. Yep. That's where I'd start if I didn't even know enough to ask questions using the correct terminology. I always skip the first 3 chapters in any technical book because I figure the authors put them in just to fill out their commitment to the publisher so he can charge more for the book -- even when the book is available for free. I figure, why learn the foundation of a new technology when there are always mailing lists manned by volunteers waiting at my beck and call -- my time is worth more than theirs. The one thing I can't figure out is why everybody keeps adding me to their MUA kill lists... Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 The things I could write here... We want instant gratification and want to drive the car without knowing anything except where the gas goes. There are plenty of Canned Asterisks for folks who don't want to bother with details like installation and dialplans. Is it easier to read 600 pages or 600 Flames? -- _ -- 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
Re: [asterisk-users] Dial plan help
You may check these videos too: http://www.youtube.com/watch?v=H1j5OrgL1og http://www.youtube.com/watch?v=7kNYuqOrP3w I find it useful, although I use visual dial plan rather than hand coding the dial plan. Either way you need to understand at least basics of asterisk dial plan structure. Rayan On 10/25/2010 7:55 PM, Jigar Joshi wrote: Ok Thanks Guys. Can you guyz suggest me upto which chapters orwhat are the chapters I should cover for my requirement. Because Its too long book :P On Mon, Oct 25, 2010 at 10:54 PM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nile Kaledon *Sent:* Monday, October 25, 2010 12:06 PM *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Dial plan help Hi Jigar, I use visual dialplan too. Nice tool. Here you can find some dial plan examples and tutorials that may help you: codezone.apstel.com http://codezone.apstel.com Nile I’ll have to agree that VDP is a nice tool, but it is just that – a tool. If you don’t know how the dialplan and commands work, it will eventually dig you into a hole you won’t get out of. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
Hi, I just downloaded your vdp file and it's working fine on my installation (Asterisk 1.4). Can you be more specific on the issue you experienced? Nile -- _ -- 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
Re: [asterisk-users] Dial plan help
Hi Jigar I am facing issue while generating a dial plan for the following case: all caller should be asked a code to enter than All the callers should be connected one extension. Try DISA component, and then use MeetMe component if you want callers to go to conference or Dial component if you want them to go to extension. I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text version for the same . Visual dialplan outputs standard extensions.conf code. You can get the code by selecting Local deploy option at preferences window or SSH to Asterisk server and check extensions.conf. I was coding dial plans in vi for some time and then switch to Visual Dialplan, much easier and faster, very useful tool. Rayan -- _ -- 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
Re: [asterisk-users] Dial plan help
I totally agree with Steve's wise advice. One should at least give himself a week learning asterisk fundamentals and related Linux basics before jumping into creating dialplans or setting up Telecom systems. Asterisk's official book's first few chapters cover all the basics which every asterisk user must to know. Otherwise seeking help here won't help because you won't be able to even understand the answers here. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-24 7:59 AM, Rayan Smith rayan.o.sm...@gmail.com wrote: Hi Jigar I am facing issue while generating a dial plan for the following case: all caller should be as... Try DISA component, and then use MeetMe component if you want callers to go to conference or Dial component if you want them to go to extension. I have created a dial plan using vdp I tried submitting it here but I don't know how to extract t... Visual dialplan outputs standard extensions.conf code. You can get the code by selecting Local deploy option at preferences window or SSH to Asterisk server and check extensions.conf. I was coding dial plans in vi for some time and then switch to Visual Dialplan, much easier and faster, very useful tool. Rayan -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial plan help
Jigar Joshi wrote: Currently I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text version for the same . After Googling a bit, I found that VDP is Visual Dial Plan for Asterisk. Neat little application, but I doubt you'll find many if any here using it. I also don't agree with their statement: Why should I use Visual Dialplan? Simply because this is the easiest and fastest way to create Asterisk dialplan. You do not need to have Asterisk dialplan development experience to create large and complex dialplans. Simply drag, drop and connect components to create the dialplan If you don't have dial plan experience, then when things aren't working, you'll be completely lost. Hence as you are now. I'd suggest you visit http://asteriskdocs.org http://www.voip-info.org And learn how to code a basic dial plan. You'll find many here willing to help you at that point. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] Dial plan help
On Mon, 18 Oct 2010, Jigar Joshi wrote: @Gilles here are my requirement.can you please help me . On Mon, 18 Oct 2010, Steve Edwards wrote: Are you putting this out to bid or are you just too lazy to read ATFOT (http://downloads.oreilly.com/books/9780596510480.pdf)? On Sat, 23 Oct 2010, Jigar Joshi wrote: I am facing issue while generating a dial plan for the following case: all caller should be asked a code to enter than All the callers should be connected one extension. also tell me testing scenario : I have pbx setup and currently I have soft phones to use as extension. Currently I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text version for the same . I have deployed that dial plan to my local system and when I dial any extension call just gets ended. At the end I should be able to dial a no from a soft phone and it should ask me a code then I should be connected to an fixed extension. Have you read the book? Have you tried some of the examples? This visual dial plan stuff is leading down a path where you will never understand how a dial plan really works and you will not learn how to help yourself -- which you will need to do since nobody on this list seems to know anything about vdp. I'm still trying to understand what you are trying to accomplish. It sounds like you want to allow callers to join a conference after entering a PIN. If so, search the book for examples on using the meetme() application. Google will also prove to be a valuable resource. It also sounds like you haven't mastered even calling from one extension to another. Learn to walk before you try to run. please also mention how to deploy sound file to system using web interface. Doesn't your vdp stuff automagically do this for you? Skip looking for some magic visual or web based tools and learn to use the Unix command line -- it's really not all that difficult. If you don't want to invest the time to learn to use the proper tools, please hire someone to do it for you. Do you repair your own [kitchen appliances|plumbing|car|computer]? You can learn any of these skills or you can hire somebody to do it for you. Do you have the basic Unix skills to use cp (from a USB stick), scp, or ftp? Try this approach: 1) Learn enough Unix to log in and edit the Asterisk configuration files using an editor like emacs, vi, or joe. 2) Create a simple dial plan so you can dial a number and play a file like demo-congrats. 3) Add to your dialplan so you can dial another number and dial another phone. 4) Add to your dialplan so you can dial a number and execute the meetme() application. At each step, observe the console output from Asterisk so you will learn what a normal call looks like and you will see useful messages that will clue you in when something doesn't work as expected. Everybody on this list is interested in helping you succeed with Asterisk, but only if you are willing to invest the effort. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
I am reattaching the file one of the svg file is pending for moderator's approval, but I am here attaching vdp file. On Thu, Oct 21, 2010 at 3:13 PM, Jigar Joshi jiga...@gmail.com wrote: It seems to have some server configuration with it, Its not getting parsed if i stop server. I am attaching svg format now On Thu, Oct 21, 2010 at 2:59 PM, Steve Howes steve-li...@geekinter.netwrote: On 21 Oct 2010, at 10:16, Jigar Joshi wrote: I have attached the dial plan file. In what format? S -- _ -- 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 first.vdp Description: Binary data -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
I think you will have better response if you can provide the actual dialplan text file, instead of the format you attached. Regards HASSAN On 2010-10-22, Jigar Joshi jiga...@gmail.com wrote: I am reattaching the file one of the svg file is pending for moderator's approval, but I am here attaching vdp file. On Thu, Oct 21, 2010 at 3:13 PM, Jigar Joshi jiga...@gmail.com wrote: It seems to have some server configuration with it, Its not getting parsed if i stop server. I am attaching svg format now On Thu, Oct 21, 2010 at 2:59 PM, Steve Howes steve-li...@geekinter.netwrote: On 21 Oct 2010, at 10:16, Jigar Joshi wrote: I have attached the dial plan file. In what format? S -- _ -- 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 -- Sent from my mobile device -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
On Thu, Oct 21, 2010 at 02:46:16PM +0530, Jigar Joshi wrote: Here I am expecting to be configured following scenario: User calls : it will play a sound will ask for input DTMF, then call will be given to particular extension for any DTMF entered. But its not working as expected. I have attached the dial plan file. Sorry, but I can't parse that VDP (Visual DialPlan?). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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
Re: [asterisk-users] Dial Plan Conf
On 21 Oct 2010, at 10:16, Jigar Joshi wrote: I have attached the dial plan file. In what format? S -- _ -- 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
Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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Re: [asterisk-users] Dial plan question.
Thanks Steve, I corrected spelling that but still having issue :-)
Issue:
when some one calls bob, I want asterisk to add @DOMAIN and make the call.
but it is not working .
--
Config files:
sip.conf
[ext-sip]
type=friend
context=phones
qualify=yes
host=external.proxy.com
extensions.conf
exten = bob,1,Dial(SIP/${ext...@ext-sip,20)
the call is not working,
log says:
chan_sip.c:5344 create_addr:no such host: ext-sip
app_dail.c:1745 unable to create channel of type 'SIP' (cause 20-unknown)
can u please correct me what I am missing
From: Steve Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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Re: [asterisk-users] Dial plan question.
Do you mean you want
exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user ext-sip to that system's extension bob?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 28, 2010, at 10:50 AM, Aditya Kumar wrote:
Thanks Steve, I corrected spelling that but still having issue :-)
Issue:
when some one calls bob, I want asterisk to add @DOMAIN and make the call.
but it is not working .
--
Config files:
sip.conf
[ext-sip]
type=friend
context=phones
qualify=yes
host=external.proxy.com
extensions.conf
exten = bob,1,Dial(SIP/${ext...@ext-sip,20)
the call is not working,
log says:
chan_sip.c:5344 create_addr:no such host: ext-sip
app_dail.c:1745 unable to create channel of type 'SIP' (cause 20-unknown)
can u please correct me what I am missing
From: Steve Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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Re: [asterisk-users] Dial plan question.
Try: exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20) ?
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Re: [asterisk-users] Dial plan question.
Thanks a lot Jim and Ryan.
It worked with changing the order as you suggested.
--
Few more questions on Dial plan:
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding Domain and call is been
sent to the other party -- Works, no media...
For the cases when it is talking to the external work,
I want Astersik not to do anything with the SDP/payload.
I want it to send as it is to the external proxy.
How can I achieve this? so that the SDP/payload will not be modified for users
talking to the external world.
I want media for those external devices to come Directly to the users in my
pbx.
Do
you mean you want
exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user
ext-sip to that system's extension bob?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
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Re: [asterisk-users] Dial plan question.
On Tue, Apr 27, 2010 at 8:48 PM, Aditya Kumar adityakumar...@yahoo.comwrote:
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this
done using extensions.conf)
my expected result is :
al...@pbx.com should be able to call b...@pbx.com
where pbx.com is astersik.
Can you pl let me know how I can achieve this?
You would need to setup each user in sip.conf like so:
[alice]
type=friend
context=alpha-names
fromuser=alice
secret=password
domain=pbx.com
[bob]
type=friend
context=alpha-names
fromuser=bob
secret=password
domain=pbx.com
etc etc..
Then in your extensions.conf, you would setup:
[alpha-names]
; Dial by name
exten = alice,1,Verbose(Calling alice)
exten = alice,n,Dial(SIP/alice,20)
exten = alice,n,Hangup()
exten = bob,1,Verbose(Calling bob)
exten = bob,n,Dial(SIP/bob,20)
exten = bob,n,Hangup()
etc etc. You could also use pattern matching in your extensions.conf like
this:
[alpha-names]
;Dial by name, pattern matching
exten = _.,1,Verbose(Calling ${EXTEN})
exten = _.,n,Dial(SIP/${EXTEN},20)
exten = _.,n,Hangup()
except that's going to catch everything, including the built-in 'h', 'i',
and 't' extensions (you can look these up on voip-info.org for more info on
those).
Configure each of your softphone clients with the usernames you defined in
your sip.conf (i.e the softphone on Alice's computer would have a username
of alice, password of password, and domain of pbx.com, using the asterisk
server as your registrar / proxy server address, same with Bob's softphone).
Your softphone has to allow alpha dialing from contacts though. You haven't
mentioned which softphone you're using, if you do that we may be able to
give you specifics for that softphone as well.
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Dial plan question.
I am not sure what your problem is. You can have a numeric extension dial an alphabetic sip user. exten = 123,1,Dial(SIP/somename) The soft phone registers to your box with whatever username you set up. If your phone can dial alpha then you can have exten = alpha,1,Dial(SIP/$(EXTEN}) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote: Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : al...@pbx.com should be able to call b...@pbx.com where pbx.com is astersik. Can you pl let me know how I can achieve this? -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply. My issue is : there is no numbers involved. I have soft clients. when a user (bob) calls Alex, he just opens his sip client and types in a...@pbx.com so how can I write the translations for a case like that? the examples you gave are when there are numbers..can u pl give me complete numbering plam From: Jim Dickenson dicken...@cfmc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, April 27, 2010 7:09:45 PM Subject: Re: [asterisk-users] Dial plan question. I am not sure what your problem is. You can have a numeric extension dial an alphabetic sip user. exten = 123,1,Dial(SIP/somename) The soft phone registers to your box with whatever username you set up. If your phone can dial alpha then you can have exten = alpha,1,Dial(SIP/$(EXTEN}) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote: Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : al...@pbx.com should be able to call b...@pbx.com where pbx.com is astersik. Can you pl let me know how I can achieve this? -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dial plan question.
here is the dail plan I am using:
my extensions file:
[globals]
[ext-sip]
host=provider.sip.com
[default]
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
expected dialing plan:
when some one calls bob,
Asterisk should add b...@provider.sip.com and sent to the external world.
But that is not working,.
can you pl let me know what I am missing?
Also,
is there a way that Asterisk will read completely b...@provider.sip.com from
the received sip message and forwards directly to that domain.
That means,
When we receive a Request to b...@provider.sip.com, Asterisk should send that
to the outgoing interface to b...@provider.sip.com\.
some plan like..
extern=b...@x.com,1,Dial(SIP/{EXTERN},20)...
From: Aditya Kumar adityakumar...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, April 27, 2010 10:11:16 PM
Subject: Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply.
My issue is :
there is no numbers involved. I have soft clients.
when a user (bob) calls Alex,
he just opens his sip client and types in a...@pbx.com
so how can I write the translations for a case like that?
the examples you gave are when there are numbers..can u pl give me complete
numbering plam
From: Jim Dickenson dicken...@cfmc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, April 27, 2010 7:09:45 PM
Subject: Re: [asterisk-users] Dial plan question.
I am not sure what your problem is. You can have a numeric extension dial an
alphabetic sip user.
exten = 123,1,Dial(SIP/somename)
The soft phone registers to your box with whatever username you set up.
If your phone can dial alpha then you can have
exten = alpha,1,Dial(SIP/$(EXTEN})
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote:
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this
done using extensions.conf)
my expected result is :
al...@pbx.com should be able to call b...@pbx.com
where pbx.com is astersik.
Can you pl let me know how I can achieve this?
--
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Re: [asterisk-users] Dial Plan configuration in asterisk
- Chandrakant Solanki solanki.chandrak...@gmail.com wrote:
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10 digits for
example like accessing an IVR menu.
Warm Regards
Venugopal G
*
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
--
Thanks, Phil
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Re: [asterisk-users] Dial Plan configuration in asterisk
19 feb 2010 kl. 11.47 skrev --[ UxBoD ]--:
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
UxBoD - you really have to read the security advisory before sending out such
examples on the mailing list. Please go to http://www.asterisk.org now.
Have a nice weekend!
Thanks,
/O
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Re: [asterisk-users] Dial Plan configuration in asterisk
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10 digits
for example like accessing an IVR menu.
Warm Regards
Venugopal G
*
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Hi
Try this
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},20)
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Chandrakant Solanki
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Re: [asterisk-users] Dial Plan configuration in asterisk
Hi
I tried the below expression.However it didn't work.
I got the below error message on my CLI
app_dial.c:871 wait for answer:Unable to forward voice or dtmf
pbx.c:3897 _ast_pbx_run: Timeout, but no ruke 't' in context 'Internal'
Warm Regards
Venugopal G
*
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Chandrakant Solanki
Sent: Friday, February 19, 2010 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Plan configuration in asterisk
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than
10 digits for example like accessing an IVR menu.
Warm Regards
Venugopal G
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Hi
Try this
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},20)
--
Regards,
Chandrakant Solanki
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Re: [asterisk-users] Dial Plan Application(main-menu)
Do you have the main-menu sound file in the correct format? Goksie On 11/20/09, Steve Edwards asterisk@sedwards.com wrote: On Fri, 20 Nov 2009, aster...@opensourcesolution.in wrote: the problem is that when call comes it answers but backgroup main menu dosent play,for test purpose i had done The problem is that you do not have (or have not provided) sufficient information to solve today's problem. You should bump up logging (logger.conf, console = debug,dtmf,error,event,notice,verbose,warning) and contemplate (for a very long time) the meaning of the messages. There are resources available on the Internet (google.com, voip-info.org) where you can find answers faster and without annoying the hell out of the list as you attempt to have others write your dialplan line-by-line, day-by-day. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Application(main-menu)
On Fri, 20 Nov 2009, aster...@opensourcesolution.in wrote: the problem is that when call comes it answers but backgroup main menu dosent play,for test purpose i had done The problem is that you do not have (or have not provided) sufficient information to solve today's problem. You should bump up logging (logger.conf, console = debug,dtmf,error,event,notice,verbose,warning) and contemplate (for a very long time) the meaning of the messages. There are resources available on the Internet (google.com, voip-info.org) where you can find answers faster and without annoying the hell out of the list as you attempt to have others write your dialplan line-by-line, day-by-day. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan sample for detecting Voice Mail
Did you #include extensions_additional.conf in your extensions.conf file?
Verify this by doing dialplan show macro-screen from CLI.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B.
Reddy Bynagari
Sent: Wednesday, August 19, 2009 2:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Dial plan sample for detecting Voice Mail
Hi,
I am trying to implement a macro-screen mentioned at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
I put the following code in my extensions_additional.conf
screen-from: You have a call;
screen-accept: Press 1 to accept this call or any other key to reject.;
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,1,Playback(screen-from)
exten = s,1,Playback(${ARG1})
exten = s,1,Read(ACCEPT|screen-accept|1)
exten = s,1,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
exten = s,1(yes),SetVar(MACRO_RESULT=CONTINUE)
exten = s,1(no),System(/bin/rm ${ARG1})
; end of [macro-screen]
[multi-dir-callback]
include = multi-dir-callback-custom
exten = _X.,1,Macro(screen,)
exten = _X.,1,Answer
exten = _X.,n,Playback(beep)
;exten =
_X.,n,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CALL_I
D}num1=${num1}CallStatus=${DIALSTATUS}state=${STATE})a
exten =
_X.,n,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CALL_I
D}num1=${num1}state=${STATE})
exten = _X.,n,Goto(${EXTEN},1)
exten =
hangup,1,DeadAGI(agi://127.0.0.1/callback_handler?num2=${EXTEN}callid=${CAL
L_ID}num1=${num1}state=${STATE})
; end of [multi-dir-callback]
It is not even recognizing the Screen macro? What I am I doing wrong?
Thanks
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Re: [asterisk-users] Dial Plan Issues
the following two lines exist in the extensions_additional.conf [from-max]exten = _X,1,Answerexten = _X,n,Queue(8000,tr,,) and it DOES exist in the output of the 'show dialplan' [ Context 'from-max' created by 'pbx_config' ] '_X' = 1. Answer() [pbx_config]2. Queue(8000|tr||) [pbx_config] yet my system doesn't use it to route regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 28 Sep 2008 23:31:46 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
On Mon, 2008-09-29 at 14:51 +, Tariq .. wrote: the following two lines exist in the extensions_additional.conf [from-max] exten = _X,1,Answer exten = _X,n,Queue(8000,tr,,) and it DOES exist in the output of the 'show dialplan' [ Context 'from-max' created by 'pbx_config' ] '_X' = 1. Answer() [pbx_config] 2. Queue(8000|tr||) [pbx_config] yet my system doesn't use it to route regards Tariq-- Maybe I missed a message or something, but I don't see a response to Tzafrir's request to see /etc/asterisk/extensions.conf. extensions_additional.conf is not extensions.conf; and unless extensions.conf includes it, it will never be a part of your dialplan. You did mention that you were using trixbox in your original question, so we referred you to a trixbox mailing list, because rumors have it that trixbox does complicated things in their dialplan to accomplish their goals, and most folks in this mailing list (but not all) don't play much with trixbox. But if you are not using trixbox, then you might look in your extensions.conf to answer these questions. Another resource you have to investigate the dialplan is in the CLI of asterisk; you can say dialplan show, or dialplan show from-max to see if the from-max context has been included. when the pbx_config module (module load pbx_config.so) loads, it will read in /etc/asterisk/extensions.conf; if it is not there, that module will not complete the loading process. If want us to evaluate why your dialplan is not working, show us the dialplan in extensions.conf. murf __ Date: Sun, 28 Sep 2008 23:31:46 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Issues On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
no.. it's directly connected to the internet.. it's not an issue of accepting
calls.. see.. the problem is the call gets to the server.. the server tries to
route it..
but as if the dial plan is not there.. it rejects the call because it doesn't
know what to do with it..
for example of my SIP.Conf
[5003]
type=peer
qualify=yes
port=5060
nat=yes
host=HOSTIP
allow=all
dial=SIP/5003
context=from-smarttech
canreinvite=no
call-limit=50
deny=0.0.0.0/0.0.0.0
permit=HOSTIP/255.255.255.255
Extensions.conf
[from-smarttech]
exten = fax,1,Goto(ext-fax,in_fax,1)
exten = s,n,Set(__FROM_DID=${EXTEN})
exten = s,n,Gosub(app-blacklist-check,s,1)
exten = s,n,GotoIf($[ ${CALLERID(name)} != ] ?cidok)
exten = s,n,Set(CALLERID(name)=${CALLERID(num)})
exten = s,n(cidok),Noop(CallerID is ${CALLERID(all)})
exten = s,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}})
exten = s,n,SetCallerPres(allowed_not_screened)
exten = s,n,Goto(ext-queues,8004,1)
let's say smarttech is a voip provider.. which forwards calls to my user on
their system .. now my server is supposed to route those calls according to the
dial plan..
the same exact settings worked like magic on another server.. but on this
server.. it just as if the context and the dial plan does not exist.!!!
any idea?
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Fri, 26 Sep 2008 11:55:45 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [asterisk-users] Dial Plan Issues
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of
my servers is not allowing the other to send calls while it is possible in the
opposit direction..
i have the same exact settings for the extensions.conf
i tried with another friend of mine.. and connected to his server.. and it
didn't allow him to send me calls..
so my question is..
is it possible that my server is not accepting any context ? it only runs the
ones that come default with Trixbix.. like chanspy, ext-local, from-trunk...
and so on..
what can i do to avoide this problem?? i can't rebuild a new box this one is a
production server and i wasn't making tests.. i was connecting two of my
employer's servers with each other..
regards
Tariq--
You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in
this group who might be able to help, but your
chances are much better in that list.
murf
The server that is not accepting calls is not behind a NAT firewall by any
chance is it?
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Re: [asterisk-users] Dial Plan Issues
This is a better question asked on a Fonality list. Maybe they have a manual. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ Want to do more with Windows Live? Learn 10 hidden secrets from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns%21550F681DAD532637%215295.entry?ocid=TXT_TAGLM_WL_domore_092008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 28 Sep 2008 10:00:56 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [asterisk-users] Dial Plan Issues This is a better question asked on a Fonality list. Maybe they have a manual.Thanks,Steve Totaro On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings,i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction..i have the same exact settings for the extensions.confi tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls..so my question is..is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on..what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other..regardsAHD Tarek SawahIntegrated Digital SystemsCCNA, MCSE, RHCE, VoIPSyria: +963 944 618286USA: +1 347 562 2308_Want to do more with Windows Live? Learn 10 hidden secrets from Jamie.http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!550F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_domore_092008___-- Bandwidth and Colocation Provided by http://www.api-digital.com --AstriCon 2008 - September 22 - 25 Phoenix, ArizonaRegister Now: http://www.astricon.netasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Thanks,Steve Totaro1.888.777.18881.240.938.1212 (cell) _ See how Windows Mobile brings your life together—at home, work, or on the go. http://clk.atdmt.com/MRT/go/msnnkwxp1020093182mrt/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote: this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk read the dial plan!! What is the dialplan? ls -ld /etc/asterisk /etc/asterisk/extensions.conf And what is the contents of extensions.conf ? What is the output of 'dialplan show' from the CLI? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
Steve Murphy wrote: On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards Tariq-- You might try a trixbox users mailing list. There might be a few trixbox users hanging around in this group who might be able to help, but your chances are much better in that list. murf The server that is not accepting calls is not behind a NAT firewall by any chance is it? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Issues
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards Tariq-- You might try a trixbox users mailing list. There might be a few trixbox users hanging around in this group who might be able to help, but your chances are much better in that list. murf AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Help
Steve Alex thanks for your help. I've got it working perfectly now.
-Jon
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 24, 2008 9:22 AM
Subject: Re: [asterisk-users] Dial Plan Help
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
-- Alex
Jon Weisman wrote:
I'd like to do the following can someone guide me on how to accomplish
this?
Call comes in via PRI and tries to go out via SIP if for some reason the
ISP
is down and the call can not go out i want it to fail over and send the
same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if
it
was unable to send the call via SIP (if the first priority failed), but
if
it did go through sip then it should just hangup the call when the person
is
done speaking.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] Dial Plan Help
On Sun, Aug 24, 2008 at 8:11 AM, Jon Weisman [EMAIL PROTECTED] wrote:
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if it
was unable to send the call via SIP (if the first priority failed), but if
it did go through sip then it should just hangup the call when the person is
done speaking.
Thanks,
Jon
Jon,
This should work just fine with the correct dial syntax, after a call
ends, the exten goes to h for hangup rather then progressing further
down the priority for the initial exten.
Double check your syntax.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Thanks,
Steve Totaro
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Re: [asterisk-users] Dial Plan Help
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
-- Alex
Jon Weisman wrote:
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if it
was unable to send the call via SIP (if the first priority failed), but if
it did go through sip then it should just hangup the call when the person is
done speaking.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] dial plan help.
This is some pretty basic stuff... (someone will probably send you a RTFM) Start with the sample dialplan (make samples I think)...trace the dialplan along to understand how it works Check the wiki and then post anything that you need help with From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sydney Web Hosting Sent: July 6, 2008 8:33 PM To: Asterisk Users List Subject: [asterisk-users] dial plan help. I have a question about the following dial plan. Ring main number playback message If press 1 got to support if press 2 go to sales //Support Play message your call is important to us then ring the phone and I pickup. //Sales Play message your call is important to us then ring the phone and I pickup. but, the problem is I only have 1 staff member at the moment. So how do we set it up if I'm out of the office, or on the mobile phone and can't answer the call. How does it know to go to voice mail? Regards Jared ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial plan help.
So how do we set it up if I'm out of the office, or on the mobile phone and can't answer the call. How does it know to go to voice mail? You set it to ring for a certain duration then go to voicemail after n seconds. You'll want an incoming call to go to a context at which point you can start deciding what to do based on key presses they make. One of your key presses (1 for support) would then go to that context possibly at which point you can ring that phone for n seconds then send it to voicemail. Check the wiki, it shows how to do this. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Call Park / Call Pickup would probably be the best option for this. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Russell Handorf Enviado el: martes, 14 de agosto de 2007 19:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Dial plan suggestions Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). What I was thinking on how to do this is using some sort of call parking for the hunt group of all the phones in the house. Once the call is picked up, it then places both the SIP phone and caller into a meetme conference room. To simply join that static assigned room, one of the other phones picks up and joins that room. What I have a concern about is if we hang up, the caller can still sit there and listen in. When no phones are active, it should disconnect the caller. Does this implementation make sense? Has anyone else done something like this? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Here's some details for you all. Asterisk 1.2 Polycom 301/601 phones As for my existing dial plan, I'm considering starting from scratch. Thanks again. Gerald A wrote: Hiya, On 8/14/07, *Russell Handorf* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). Wives are the most difficult customers. They are demanding, and you can't ever get away with not making them happy. :) What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). You leave out what kind of phones you are using. It might be as easy as using a line appearance for a parking lot - or not - depending on what kind of phone you are using. SIP, ZAP and IAX phones, and even some within that may or may not support it. I have some GXP2000's, and I think I would do it that way. HTH ( a bit), Gerald. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
On Tue, 14 Aug 2007, Russell Handorf wrote: Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). Give her a DECT phone so she can carry it about with her. Worked for my wife! Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Gordon Henderson wrote: On Tue, 14 Aug 2007, Russell Handorf wrote: Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). Give her a DECT phone so she can carry it about with her. Worked for my wife! Gordon Parking is pretty easy once you try it, then no running to the other phone. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Russell Handorf wrote: Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to transfer calls to another phone; she'd rather just pick up the phone and have the call be active there as well (like good 'ol land lines). What I was thinking on how to do this is using some sort of call parking for the hunt group of all the phones in the house. Once the call is picked up, it then places both the SIP phone and caller into a meetme conference room. To simply join that static assigned room, one of the other phones picks up and joins that room. What I have a concern about is if we hang up, the caller can still sit there and listen in. When no phones are active, it should disconnect the caller. Does this implementation make sense? Has anyone else done something like this? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Anthony Francis wrote: You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Isn't this what Shared Line Appearance is supposed to do? (Supported in 1.4...) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Since I dont use 1.4 then you tell me. :) Stephen Bosch wrote: Anthony Francis wrote: You want a key system, the fianl frontier of an asterisk implementation, and currently my holy grail. The best way to do it in an ugly way is to park the call and have a speed dial for pickup. Some phones like Aastra 55i and 57i can even have their hold button reprogrammed to blind transfer to the call parking. Isn't this what Shared Line Appearance is supposed to do? (Supported in 1.4...) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Anthony Francis wrote: Since I dont use 1.4 then you tell me. :) This functionality is supposed to be supported in 1.4, though I've never personally tested it. When it's configured it gives the key system behaviour you describe. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan suggestions
Stephen Bosch wrote: Anthony Francis wrote: Since I dont use 1.4 then you tell me. :) This functionality is supposed to be supported in 1.4, though I've never personally tested it. When it's configured it gives the key system behaviour you describe. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looks like you are right, man I cannot wait for them to fix the CDR problems in 1.4 so that I can move to it. http://www.voip-info.org/wiki/view/Asterisk+SLA Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
On Mon, 2007-07-30 at 16:09 +0100, Chris Blunt wrote:
If the PSTN is in use on SPA3102 I need a way to get the call to then
route out over IAX termination.
Usually, the best way to accomplish this is to send a call to your
Linksys ATA by using the Dial application from the dialplan, and then
looking at the result that gets set in the DIALSTATUS variable. For
example, you could try something like this:
exten = 123,1,Dial(SIP/linksys/5551212,30)
exten = 123,n,GotoIf($[${DIALSTATUS} = CONGESTION]?try-iax)
exten = 123,n,Busy(3)
exten = 123,n,Hangup()
exten = 123,n(try-iax),Dial(IAX2/my_iax_peer/5551212,30)
Obviously my example isn't that robust... it's simply meant to
illustrate the idea. (It depends on the SPA3102 returning a status code
that maps to CONGESTION if it's already in use... I don't have an
SPA3102, so I can't tell you how it actually performs.)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Dial plan inquiry using GotoIf()
On Wed, 6 Jun 2007, Mike Lynchfield wrote:
yes on home pbx i love the s/CALLERID..
maybe you should
f($[${CALLERID(number)} = 15552221313]?15:5)
try to isolate string to strings.
this is not good i think
you need qhotes on the callerid part too if you evaluate to the 1555xxx
f($[${CALLERID(number)} = 15552221313]?15:5)
Just a note on dialplan programming here - I understand that the Jump to
n+101 is depreciated now, and keeping track of line numbers is something I
decided to give up on when I left BASIC programming on an Apple II, 25
years ago...
So from this:
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
to this:
exten = s,n,GotoIf($[${CALLERID(number)} = 15552221313]?trapped)
(no need for the false jump here either)
and from this:
exten = s,15,HangUp
to this:
exten = s,n(trapped),Hangup
Always check the README.variables in the docs directory too.
Gordon
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Re: [asterisk-users] Dial plan inquiry using GotoIf()
On Wed, 2007-05-30 at 20:05 -0400, Steve Finkelstein wrote:
Thanks for the help on this thread all.
It would make sense if I write an AGI and incorporate a DB backend to
check against numbers I want explicitly dropped. If anyone has such a
utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
it up and probably provide a web frontend for adding/removing numbers.
You can still use the dialplan with the DB func to check incoming
CID info. Also, the Dial() app has several options for call screening
and
privacy; these would be performed when dialing your extension.
You can have Dial keep a DB of callers, and remember whether to always
just patch them right thru, play them a polite go away and don't come
back,
or send them off to torture scripts, or just route them straight to VM.
And, Dial() will ask you what you want to do, on the first call. Read
thru the Dial doc you get with core show application dial. There's
an option to store an intro from each caller, where it records in a
sound file, who they say they are. I have several hundreds of these, and
play them as the
call comes in, so we know who's calling without having to run to a CID
display.
For those who have poor to no vision, this can be a cool feature.
murf
- sf
C F wrote:
It fails because the right function is ${CALLERID(num)}
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new
stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5) in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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