Re: [asterisk-users] help with dialplan
On Tue, Aug 31, 2010 at 9:34 AM, Danny Nicholas da...@debsinc.com wrote: Why not just copy the _1NXXNXX line into the remote context? Well, that could be done, and probably would be a good tactic if you have lots of DID's and want to do db lookup or something to direct the next call leg. But, if you only have one or two DID's, all the machinery and programming seem a bit overkill. murf -- -- Steve Murphy ParseTree Corp -- _ -- 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] help with dialplan
Here's the updated debug log. http:/www.computerworkx.net/client/Document.txt On 8/30/2010 2:55 PM, Paul Belanger wrote: On Mon, Aug 30, 2010 at 2:48 PM, Todd Reesetrees...@gmail.com wrote: Thanks for pointing out the misspelling. I've corrected that and still no luck. Create a new debug log with your recent changes, re-attach it the list. -- _ -- 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] help with dialplan
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reese trees...@gmail.com wrote: Here's the updated debug log. [Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to extension '6789542133' rejected because extension not found in context 'remote'. So, again, a context problem. You can confirm by entering: *CLI dialplan show 6789542...@remote -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] help with dialplan
asterisk*CLI dialplan show 6789542...@remote There is no existence of 'remote' context Command 'dialplan show 6789542...@remote' failed. asterisk*CLI On 8/31/2010 9:58 AM, Paul Belanger wrote: dialplan show 6789542...@remote -- _ -- 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] help with dialplan
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
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Re: [asterisk-users] help with dialplan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
Ok. I'm a late joiner to this thread. Reading the original post I see
that you are trying to do an external SIP dial to 678-954-2133. These
questions:
1. Does the dial string need to be 1xxx instead of xxx (presumably local 10
digit dialing)? If yes, change
exten = _NXXNXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
to
exten = _NXXNXX,n,Dial(SIP/v6781234567/1${EXTEN},40,Ttr)
2. voipdialACA and v6781234567 are registered trunks with credentials?
Hope this helps.
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Re: [asterisk-users] help with dialplan
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into the
right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and each
of those
include remote.
Straighten out that mess and maybe things might work. Just a guess, but
worth a try!
murf
On Tue, Aug 31, 2010 at 8:25 AM, Todd Reese trees...@gmail.com wrote:
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
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Re: [asterisk-users] help with dialplan
I had already check on this. Thanks for the info, though.
On 8/31/2010 10:36 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Subject: Re: [asterisk-users] help with dialplan
asterisk*CLI dialplan show 6789542...@remote
There is no existence of 'remote' context
Command 'dialplan show 6789542...@remote' failed.
asterisk*CLI
On 8/31/2010 9:58 AM, Paul Belanger wrote:
dialplan show 6789542...@remote
Ok. I'm a late joiner to this thread. Reading the original post I see
that you are trying to do an external SIP dial to 678-954-2133. These
questions:
1. Does the dial string need to be 1xxx instead of xxx (presumably local 10
digit dialing)? If yes, change
exten = _NXXNXX,n,Dial(SIP/v6781234567/${EXTEN},40,Ttr)
to
exten = _NXXNXX,n,Dial(SIP/v6781234567/1${EXTEN},40,Ttr)
2. voipdialACA and v6781234567 are registered trunks with credentials?
Hope this helps.
--
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Re: [asterisk-users] help with dialplan
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call from '150' to extension '16789542133' rejected because extension
not found in context 'remote'.
asterisk*CLI dialplan show 16789542...@remote
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
-= 7 extensions (7 priorities) in 7 contexts. =-
[Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper:
Avoiding circular include of from-internal within remote
On 8/31/2010 10:49 AM, Steve Murphy wrote:
Todd--
There is probably some nifty anti-infinite-recursion code in the
extensions.conf parser,
to keep asterisk from going into infinite loops trying to descend into
the right context.
In your dialplan, [remote] includes dialout1, dialout2, dialout3, and
each of those
include remote.
Straighten out that mess and maybe things might work. Just a guess,
but worth a try!
murf
On Tue, Aug 31, 2010 at 8:25 AM, Todd Reese trees...@gmail.com
mailto:trees...@gmail.com wrote:
From extensions.conf
[remote]
include = from-internal
include = dialout1
include = dialout2
include = dialout3
include = intercom
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[dialout1]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[dialout2]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[dialout3]
include = from-internal
include = 411
include = remote
exten = 911,1,Goto(nineoneone,s,1)
exten = _1NXXNXX,n,Dial(SIP/v6783747049/${EXTEN},40,Ttr)
On 8/31/2010 9:58 AM, Paul Belanger wrote:
On Tue, Aug 31, 2010 at 9:16 AM, Todd Reesetrees...@gmail.com
mailto:trees...@gmail.com wrote:
Here's the updated debug log.
[Aug 30 15:27:41] NOTICE[11568] chan_sip.c: Call from '150' to
extension '6789542133' rejected because extension not found in
context
'remote'.
So, again, a context problem. You can confirm by entering:
*CLI dialplan show 6789542...@remote
--
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asterisk-users mailing list
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--
Steve Murphy
ParseTree Corp
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Re: [asterisk-users] help with dialplan
On Tue, Aug 31, 2010 at 9:14 AM, Todd Reese trees...@gmail.com wrote:
Interesting things going on herel.
After your suggestions, Steve. I reran the dialplan show
16789542...@remote command with the below results.
Phone calls are geting the 404 error and the NOTICE on the console.
[Aug 31 11:06:46] NOTICE[11884]: chan_sip.c:20161 handle_request_invite:
Call from '150' to extension '16789542133' rejected because extension not
found in context 'remote'.
-- This one is easy to solve, Add the extension 16789542133 to the remote
context and have it do what you need to be done for an incoming call.
and also, see below:
asterisk*CLI dialplan show 16789542...@remote
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout1' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6787500600/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout2' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/voip.comACA/${EXTEN},40,Ttr)
[pbx_config]
[ Included context 'dialout3' created by 'pbx_config' ]
'_1NXXNXX' = -2. Dial(SIP/v6783747049/${EXTEN},40,Ttr)
[pbx_config]
-= 7 extensions (7 priorities) in 7 contexts. =-
[Aug 31 11:03:38] WARNING[11903]: pbx.c:5741 show_dialplan_helper: Avoiding
circular include of from-internal within remote
See the Avoiding circular include? Get rid of that by removing one of the
includes that make the cycle; make your
inclusions hierarchical. One context to include them all, and in the
darkness bind them! (sorry, too much Tolkien)
murf
--
Steve Murphy
ParseTree Corp
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Re: [asterisk-users] help with dialplan
Why not just copy the _1NXXNXX line into the remote context? -- _ -- 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
[asterisk-users] help with dialplan
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten = 44,1,Macro(oneline,${QPHONE4})
exten = 45,1,Macro(oneline,${QPHONE5})
exten = 46,1,Macro(oneline,${QPHONE6})
exten = 47,1,Macro(oneline,${QPHONE7})
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[macro-oneline]
exten = s,1,Set(CHANNEL(musicclass)=default)
exten = s,n,Dial(${ARG1},20,Ttr)
exten = s,n,Voicemail(${MACRO_EXTEN})
exten = s,n,Hangup
exten = s,102,Voicemail(${MACRO_EXTEN})
exten = s,103,Hangup
[dialout1]
include = from-internal
include = 411
exten =
Re: [asterisk-users] help with dialplan
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com wrote: I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. Post a debug log of the problem: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] help with dialplan
Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}$
{QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${
ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNET
PHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten = 44,1,Macro(oneline,${QPHONE4})
exten = 45,1,Macro(oneline,${QPHONE5})
exten = 46,1,Macro(oneline,${QPHONE6})
exten = 47,1,Macro(oneline,${QPHONE7})
exten = 150,1,Macro(oneline,${EXTERNPHONE0})
[macro-oneline]
exten = s,1,Set(CHANNEL(musicclass)=default)
exten = s,n,Dial(${ARG1},20,Ttr)
exten = s,n,Voicemail(${MACRO_EXTEN})
exten = s,n,Hangup
exten = s,102,Voicemail(${MACRO_EXTEN})
exten =
Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:
Todd
How do you have the context in the phones sip configs set?
Bryant
*
From*: Todd Reese trees...@gmail.com mailto:trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten =
Re: [asterisk-users] help with dialplan
Here's a debug for extension 150 [Aug 30 11:34:53] VERBOSE[2099] config.c: == Parsing '/etc/asterisk/logger.conf': [Aug 30 11:34:53] DEBUG[2099] config.c: Parsing /etc/asterisk/logger.conf [Aug 30 11:34:53] VERBOSE[2099] config.c: == Found [Aug 30 11:34:53] VERBOSE[2099] logger.c: Asterisk Event Logger restarted [Aug 30 11:34:53] VERBOSE[2099] logger.c: Asterisk Queue Logger restarted [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 0 [ 38]: OPTIONS sip:76.122.117.31:5060 SIP/2.0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 1 [ 44]: Via: SIP/2.0/UDP 64.34.245.174:5060;branch=0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 2 [ 38]: From: sip:pin...@voip.com;tag=7c9c6206 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 3 [ 26]: To: sip:76.122.117.31:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 4 [ 47]: Call-ID: 89c833e4-9c8f2516-5bb...@64.34.245.174 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 5 [ 15]: CSeq: 1 OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 7 [ 0]: [Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.1.102:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 89c833e4-9c8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP) [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Received OPTIONS (3) - Command in SIP OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 64.34.245.174:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be handled, bad request: 89c833e4-9c8f2516-5bb...@64.34.245.174 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 0 [ 38]: OPTIONS sip:76.122.117.31:5060 SIP/2.0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 1 [ 44]: Via: SIP/2.0/UDP 64.34.245.174:5060;branch=0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 2 [ 38]: From: sip:pin...@voip.com;tag=1f9c6206 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 3 [ 26]: To: sip:76.122.117.31:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 4 [ 47]: Call-ID: 89c833e4-3f8f2516-5bb...@64.34.245.174 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 5 [ 15]: CSeq: 1 OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Header 7 [ 0]: [Aug 30 11:34:53] DEBUG[2079] acl.c: Found IP address for this socket [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.1.102:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Allocating new SIP dialog for 89c833e4-3f8f2516-5bb...@64.34.245.174 - OPTIONS (No RTP) [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Received OPTIONS (3) - Command in SIP OPTIONS [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 64.34.245.174:5060 [Aug 30 11:34:53] DEBUG[2079] chan_sip.c: SIP message could not be handled, bad request: 89c833e4-3f8f2516-5bb...@64.34.245.174 [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog '89c833e4-806e2516-79b...@64.34.245.174' [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 89c833e4-806e2516-79b...@64.34.245.174 [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Auto destroying SIP dialog '89c833e4-236e2516-79b...@64.34.245.174' [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Destroying SIP dialog 89c833e4-236e2516-79b...@64.34.245.174 [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: --- SIP read from UDP:97.80.176.231:5060 --- - [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 0 [ 0]: [Aug 30 11:34:54] DEBUG[2079] chan_sip.c:Body 0 [ 0]: [Aug 30 11:34:54] VERBOSE[2079] chan_sip.c: --- SIP read from UDP:97.80.176.231:5060 --- INVITE sip:6789542...@qci.homeip.net SIP/2.0 Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af From: ATAP sip:1...@qci.homeip.net;tag=ee0cedf5f71d40f9 To: sip:6789542...@qci.homeip.net Contact: sip:1...@10.11.17.24:5060;transport=udp Supported: replaces, timer, path P-Early-Media: Supported Call-ID: 62f35b2ee0ada...@10.11.17.24 CSeq: 21395 INVITE User-Agent: Grandstream GXP2000 1.2.3.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 345 v=0 o=150 8000 8000 IN IP4 10.11.17.24 s=SIP Call c=IN IP4 10.11.17.24 t=0 0 m=audio 5050 RTP/AVP 0 8 4 18 2 97 9 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 - [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 0 [ 44]: INVITE sip:6789542...@qci.homeip.net SIP/2.0 [Aug 30 11:34:54] DEBUG[2079] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 10.11.17.24:5060;branch=z9hG4bK82d40955a28df1af
Re: [asterisk-users] help with dialplan
Todd
Your context must be set to where you want your extension to start each
time it dials out. Without getting into your dialplan code too much try
changing the context to point to dialout1
context=dialout1
If dialout1 is working you should be able to dial.
The best way to handle this is to create a context that when you dial from
your phones it decieds if you have dialed an extension or an external
number and then routes the call correclty. This way you can pickup an
extension and dial either and get the desired results.
Bryant
From: Todd Reese trees...@gmail.com
Sent: Monday, August 30, 2010 11:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote: Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}$
{QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${
ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNET
PHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2
Re: [asterisk-users] help with dialplan
Unfortunately, that didn't work. The phone is still giving me a 404
error.
I have my own system that is 1.6.2.7 with Grandstream phones that works
fine. Using it as a guide, I built this server for a client which also
has Grandstream phones.
Last week, it dialed out fine. Since the weekend, no dialing at all.
On 8/30/2010 11:42 AM, Bryant Zimmerman wrote:
Todd
Your context must be set to where you want your extension to start
each time it dials out. Without getting into your dialplan code too
much try changing the context to point to dialout1
context=dialout1
If dialout1 is working you should be able to dial.
The best way to handle this is to create a context that when you dial
from your phones it decieds if you have dialed an extension or an
external number and then routes the call correclty. This way you can
pickup an extension and dial either and get the desired results.
Bryant
*From*: Todd Reese trees...@gmail.com
*Sent*: Monday, August 30, 2010 11:20 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] help with dialplan
Here is the sip.conf portion for extension 150
[150]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1234567890
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote:
Todd
How do you have the context in the phones sip configs set?
Bryant
*
From*: Todd Reese trees...@gmail.com mailto:trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include
Re: [asterisk-users] help with dialplan
On Mon, Aug 30, 2010 at 11:42 AM, Todd Reese trees...@gmail.com wrote: Here's a debug for extension 150 In the future, simply attach your debug log to your email. Here is your problem: [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension '6789542133' rejected because extension not found in context 'extensions.conf'. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] help with dialplan
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com wrote:
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten = 33,1,Macro(oneline,${GMNETPHONE3})
exten = 34,1,Macro(oneline,${GMNETPHONE4})
exten = 35,1,Macro(oneline,${GMNETPHONE5})
exten = 36,1,Macro(oneline,${GMNETPHONE6})
exten = 37,1,Macro(oneline,${GMNETPHONE7})
exten = 40,1,Macro(oneline,${QPHONE0})
exten = 41,1,Macro(oneline,${QPHONE1})
exten = 42,1,Macro(oneline,${QPHONE2})
exten = 43,1,Macro(oneline,${QPHONE3})
exten = 44,1,Macro(oneline,${QPHONE4})
exten = 45,1,Macro(oneline,${QPHONE5})
exten = 46,1,Macro(oneline,${QPHONE6})
exten = 47,1,Macro(oneline,${QPHONE7})
exten =
Re: [asterisk-users] help with dialplan
I actually found that one and corrected it. I have replaced the context with the from-internal, remote, and dialout1. Each has produced the same results of a 404 error. On 8/30/2010 2:10 PM, Paul Belanger wrote: On Mon, Aug 30, 2010 at 11:42 AM, Todd Reesetrees...@gmail.com wrote: Here's a debug for extension 150 In the future, simply attach your debug log to your email. Here is your problem: [Aug 30 11:34:55] NOTICE[2079] chan_sip.c: Call from '150' to extension '6789542133' rejected because extension not found in context 'extensions.conf'. -- _ -- 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] help with dialplan
Thanks for pointing out the misspelling. I've corrected that and
still no luck.
On 8/30/2010 2:33 PM, Alex Bell wrote:
possibly check you spelling: [from-interal] - [dialout1]
include = from-internal
??
On Mon, Aug 30, 2010 at 10:14 AM, Todd Reese trees...@gmail.com
mailto:trees...@gmail.com wrote:
Hi all,
I've been have problems with getting this system on line and would
like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk is
1.6.2.11. Below is the extensions.conf file.
[globals]
QPHONE0=SIP/10
QPHONE1=SIP/11
QPHONE2=SIP/12
QPHONE3=SIP/13
QPHONE4=SIP/14
QPHONE5=SIP/15
QPHONE6=SIP/16
QPHONE7=SIP/17
ACAPHONE0=SIP/20
ACAPHONE1=SIP/21
ACAPHONE2=SIP/22
ACAPHONE3=SIP/23
ACAPHONE4=SIP/24
ACAPHONE5=SIP/25
ACAPHONE6=SIP/26
ACAPHONE7=SIP/27
GMNETPHONE0=SIP/30
GMNETPHONE1=SIP/31
GMNETPHONE2=SIP/32
GMNETPHONE3=SIP/33
GMNETPHONE4=SIP/34
GMNETPHONE5=SIP/35
GMNETPHONE6=SIP/36
GMNETPHONE7=SIP/37
EXTERNPHONE0=SIP/150
CPHONE1=SIP/1678000
CPHONE2=SIP/177
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/G1
; Change this for production use:
EMERGENCY_NUM=6789542133
[from-pstn]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn1]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn2]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn3]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn4]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming3,s,1)
[from-pstn5]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming2,s,1)
[from-pstn6]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn7]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[from-pstn8]
exten = s,1,Set(FROM_DID=678000)
exten = s,n,NoOp(id is ${FROM_DID})
exten = s,n,Goto(incoming1,s,1)
[incoming1]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${QPHONE0}${QPHONE1}${QPHONE2}${QPHONE3}${QPHONE4}${QPHONE5}${QPHONE6}${QPHONE7},40,Ttr)
exten = s,n,Hangup
[incoming2]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten =
s,n,Dial(${ACAPHONE0}${ACAPHONE1}${ACAPHONE2}${ACAPHONE3}${ACAPHONE4}${ACAPHONE5}${ACAPHONE6}${ACAPHONE7},40,TTr)
exten = s,n,Hangup
[incoming3]
include = from-internal
include = parkedcalls
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,Set(CHANNEL(musicclass)=QCI)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Background(thank-you-for-calling)
exten = s,n,GotoIfTime(17:01-09:00|*|*|*?closed,s,1)
exten =
s,n,Dial(${GMNETPHONE0}${GMNETPHONE1}${GMNETPHONE2}${GMNETPHONE3}${GMNETPHONE4}${GMNETPHONE5}${GMNETPHONE6}${GMNETPHONE7},40,Ttr)
exten = s,n,Hangup
[from-interal]
include = dialout1
include = dialout2
include = dialout3
include = parkedcalls
include = intercom
exten = 10,1,Macro(oneline,${QPHONE0})
exten = 11,1,Macro(oneline,${QPHONE1})
exten = 12,1,Macro(oneline,${QPHONE2})
exten = 13,1,Macro(oneline,${QPHONE3})
exten = 14,1,Macro(oneline,${QPHONE4})
exten = 15,1,Macro(oneline,${QPHONE5})
exten = 16,1,Macro(oneline,${QPHONE6})
exten = 17,1,Macro(oneline,${QPHONE7})
exten = 20,1,Macro(oneline,${ACAPHONE0})
exten = 21,1,Macro(oneline,${ACAPHONE1})
exten = 22,1,Macro(oneline,${ACAPHONE2})
exten = 23,1,Macro(oneline,${ACAPHONE3})
exten = 24,1,Macro(oneline,${ACAPHONE4})
exten = 25,1,Macro(oneline,${ACAPHONE5})
exten = 26,1,Macro(oneline,${ACAPHONE6})
exten = 27,1,Macro(oneline,${ACAPHONE7})
exten = 30,1,Macro(oneline,${GMNETPHONE0})
exten = 31,1,Macro(oneline,${GMNETPHONE1})
exten = 32,1,Macro(oneline,${GMNETPHONE2})
exten =
Re: [asterisk-users] help with dialplan
On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese trees...@gmail.com wrote: Thanks for pointing out the misspelling. I've corrected that and still no luck. Create a new debug log with your recent changes, re-attach it the list. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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
[asterisk-users] help with dialplan
I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working [smvoice-sip] exten = 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 1044,n,Hangup ; changing 1044 to 10 works find. [smvoice-sip] exten = 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 10,n,Hangup I am running 1.4.22 and DAHDI 2.0.0 complete. Why is it picking up 10 when trying to dial 1044. How can I determine what is going on here. Thanks, Jerry This is the SIP debug for the 1044 case that does not work. - Use 'exit' when done Asterisk 1.4.22, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [0mConnected to Asterisk 1.4.22 currently running on devcentos5x64 (pid = 3127) devcentos5x64*CLI Verbosity is at least 5 [Kdevcentos5x64*CLI --- SIP read from 192.168.1.89:5060 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 - --- (14 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (no NAT) to 192.168.1.89:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D;received=192.168.1.89 From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone;tag=as5a3d998e Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces? Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1f1b706f Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user '404' ?? --- SIP read from 192.168.1.89:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bKb872214aDBECCC5D From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone;tag=as5a3d998e CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Max-Forwards: ?70 Content-Length: 0 - --- (11 headers 0 lines) --- ? [Kdevcentos5x64*CLI --- SIP read from 192.168.1.89:5060 --- INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.89;branch=z9hG4bK9c456f3360D09552 From: 404 sip:[EMAIL PROTECTED];tag=25AB8538-7BACFE71 To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=404, realm=asterisk, nonce=1f1b706f, uri=sip:[EMAIL PROTECTED];user=phone, response=c6e14f94fa0bbe3d742b6f570982ed79, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 11 lines) --- Sending to 192.168.1.89 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user '404' Found RTP audio
Re: [asterisk-users] help with dialplan
Jerry Geis wrote: I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working [smvoice-sip] exten = 1044,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 1044,n,Hangup ; changing 1044 to 10 works find. [smvoice-sip] exten = 10,1,Dial(SIP/devcentos5x64_to_bt610tmm/1044) exten = 10,n,Hangup I am running 1.4.22 and DAHDI 2.0.0 complete. Why is it picking up 10 when trying to dial 1044. How can I determine what is going on here. Thanks, Jerry debug snipped Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? ___ -- 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] help with dialplan
Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? From you message I tried a couple things... Clicking New call, then starting to dial this is when it messes up. when I start entering the number first then click dial this successfull does the 1044 and I am connected as I thought. How do I turn off this overlap dial? Thanks so much. jerry ___ -- 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] help with dialplan
Jerry Geis wrote: How do I turn off this overlap dial? You need to review the dialing rules for the Polycoms. They'd be located in the ftp directory that you've setup for your Polycoms to pull their configs from. It's located in the sip.cfg. Look for the line: digitmap dialplan.digitmap= 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dialplan
On Fri, Nov 7, 2008 at 11:20 AM, Brent Davidson [EMAIL PROTECTED] wrote: Jerry Geis wrote: Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? From you message I tried a couple things... Clicking New call, then starting to dial this is when it messes up. when I start entering the number first then click dial this successfull does the 1044 and I am connected as I thought. How do I turn off this overlap dial? Thanks so much. jerry It's been a while since I've used a polycom so I'm trying to look it up. From what I can see the automatic dialing in Polycoms is accomplished with the digitmap setting. Any of the patterns set in digitmap are dialed automatically as soon as one is recognized. You can try removing everything from the digitmap to force users to click dial on every call. You could do that, or you could read the extensive writeups on www.voip-info.org and figure out a phone dialplan that works for you. That would be my long term suggestion. I try to replicate a POTS line as much as possible, or at least an office phone, with 9 to get out since most people are already hard wired for that in an office environment. The last thing you need is someone trying to dial 911 or whatever your emergency number is and in panic, forgetting to press dial. It isn't that hard to understand, and I was forced to since different regions have seven digit dialing but it is all ten or eleven in the Maryland area. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] help with dialplan
On Fri, Nov 7, 2008 at 11:27 AM, Doug Lytle [EMAIL PROTECTED] wrote: Jerry Geis wrote: How do I turn off this overlap dial? You need to review the dialing rules for the Polycoms. They'd be located in the ftp directory that you've setup for your Polycoms to pull their configs from. It's located in the sip.cfg. Look for the line: digitmap dialplan.digitmap= Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. For two phones, I would just use the web interface.. That is of course if you plan on keeping a small amount of phones. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] help with dialplan
Steve Totaro wrote: For two phones, I would just use the web interface.. That is of course if you plan on keeping a small amount of phones. Or, if you absolutely hate the web interface :-P -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with dialplan
Jerry Geis wrote: Are your polycom phones set up for overlap dialing or do you dial the number then press a key to dial? From you message I tried a couple things... Clicking New call, then starting to dial this is when it messes up. when I start entering the number first then click dial this successfull does the 1044 and I am connected as I thought. How do I turn off this overlap dial? Thanks so much. jerry It's been a while since I've used a polycom so I'm trying to look it up. From what I can see the automatic dialing in Polycoms is accomplished with the digitmap setting. Any of the patterns set in digitmap are dialed automatically as soon as one is recognized. You can try removing everything from the digitmap to force users to click dial on every call. ___ -- 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] Help with Dialplan Rules Please!
Thanks Alex, it looks like you had a great answer to the issue at hand. On 10/17/06, Alex Robar [EMAIL PROTECTED] wrote:If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it should work for you. Both outbound routes can use the same trunk without issue. AlexOn 10/17/06, Chris Ramsey [EMAIL PROTECTED] wrote: This was posted at The Asterisk Blog Forums Click here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with Dialplan Rules Please!
This was posted at The Asterisk Blog ForumsClick here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with Dialplan Rules Please!
If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it should work for you. Both outbound routes can use the same trunk without issue. AlexOn 10/17/06, Chris Ramsey [EMAIL PROTECTED] wrote: This was posted at The Asterisk Blog Forums Click here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with dialplan
Cosmin Prund wrote: I've got a Mobile-to-PBX gateway installed and I want the ability to dial from my mobile phone into my PBX and next dial a land-line from the PBX so I can make cheep mobile-to-land-line calls while on the go. I've contemplated using the WaitExten application but it only seems to wait for ONE digit! Is there a way to put the calling mobile phone into a context and wait for a full-length extension? The documentation on voip-info has just the example you need. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+WaitExten -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with dialplan
I've got a Mobile-to-PBX gateway installed and I want the ability to dial from my mobile phone into my PBX and next dial a land-line from the PBX so I can make cheep mobile-to-land-line calls while on the go. I've contemplated using the WaitExten application but it only seems to wait for ONE digit! Is there a way to put the calling mobile phone into a context and wait for a full-length extension? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA
On Tue, Nov 08, 2005 at 07:38:20AM -0500, Frank Tarczynski exclaimed: Since this is my DID, I want the line to ring as normal but allow a user to breakout and ultimately get an outgoing line. In this way the DID line would function as a normal telephone line. A point lost on many responders! I don't want to have to go into voicemail to breakout since I don't want to give voicemail access to some of the people I will give targeted outgoing access to. This snippet from extensions.conf seem to work OK for internal extensions. Changing the context appears to stop the Playtones() OK. Any reasons why I shouldn't turn it lose? [incoming] exten = 1004,1,Playtones(ring) exten = 1004,2,Waitexten(20) exten = 1004,3,StopPlaytones exten = 1004,4,Goto(incoming,1002,1) exten = *,1,Goto(disa-1,s,1) [disa-1] exten = s,1,Playback(enter pin) exten = s,2,ResponseTimeout(20) exten = s,3,DigitTimeout(5) exten = s,4,DISA(no-password|outgoing) exten = s,5,Congestion Whats the ip address of this system? Just kidding, but your should seriously consider an authenticate() before DISA. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with dialplan to allow breakout to DISA
Since this is my DID, I want the line to ring as normal but allow a user to breakout and ultimately get an outgoing line. In this way the DID line would function as a normal telephone line. A point lost on many responders! I don't want to have to go into voicemail to breakout since I don't want to give voicemail access to some of the people I will give targeted outgoing access to. This snippet from extensions.conf seem to work OK for internal extensions. Changing the context appears to stop the Playtones() OK. Any reasons why I shouldn't turn it lose? [incoming] exten = 1004,1,Playtones(ring) exten = 1004,2,Waitexten(20) exten = 1004,3,StopPlaytones exten = 1004,4,Goto(incoming,1002,1) exten = *,1,Goto(disa-1,s,1) [disa-1] exten = s,1,Playback(enter pin) exten = s,2,ResponseTimeout(20) exten = s,3,DigitTimeout(5) exten = s,4,DISA(no-password|outgoing) exten = s,5,Congestion Message: 21 Date: Mon, 7 Nov 2005 14:25:50 -0500 (EST) From: Frank Tarczynski [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Help with dialplan to allow breakout to DISA To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;charset=iso-8859-1 Yes, I know. BUT, I want the line to work as normal for incoming calls AND allow the user to breakout. So how do I merge: [incoming] exten = 1000,1,Ringing exten = 1000,2,Answer exten = 1000,n,Dial(IAX,iaxy/20) exten = 1000,n,Voicemail() exten = 1000,n,Hangup AND exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) exten = *, 3, Hangup to have Asterisk answer the line as normal but also react to the user pressing *? I've tried putting' all of the above in the same context but it doesn't work when I call in and press *. Frank Message: 10 Date: Mon, 7 Nov 2005 12:45:05 -0500 From: Rusty Dekema [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I do it this way: exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) exten = *, 3, Hangup It seems to work fine... -Rusty On 11/7/05, Frank Tarczynski [EMAIL PROTECTED] wrote: I'm trying to set-up a dialplan for incoming calls that allows a breakout by pressing something like *. Users would then be able to get an inside dial tone for voicemail, outgoing calls, etc. I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck. Are there any good documents out there to assist me in this? Frank --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with dialplan to allow breakout to DISA
I'm trying to set-up a dialplan for incoming calls that allows a breakout by pressing something like *. Users would then be able to get an inside dial tone for voicemail, outgoing calls, etc. I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck. Are there any good documents out there to assist me in this? Frank ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA
I do it this way: exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) exten = *, 3, Hangup It seems to work fine... -Rusty On 11/7/05, Frank Tarczynski [EMAIL PROTECTED] wrote: I'm trying to set-up a dialplan for incoming calls that allows a breakoutby pressing something like *.Users would then be able to get an insidedial tone for voicemail, outgoing calls, etc.I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck.Are there any good documents out there to assist me in this?Frank___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
