[Asterisk-Users] direct-inward-dialing (DID)

[john lawler]

I know that Asterisk supports DID, but does anyone have documentation on
how to write the configuration for it?

I'll be trying to setup a hybrid system where some incoming numbers will
be DID enabled and others won't, so I'll need to be able to sort between
the two, i.e. directly connect the DID dialed numbers and route the
others to an autoattendant for extension dialing.

Thanks,

John Lawler

[Steven Critchfield]

Post by john lawler
I know that Asterisk supports DID, but does anyone have documentation on
how to write the configuration for it?
I'll be trying to setup a hybrid system where some incoming numbers will
be DID enabled and others won't, so I'll need to be able to sort between
the two, i.e. directly connect the DID dialed numbers and route the
others to an autoattendant for extension dialing.

Dude, get with the reading and googling. You sent this question less
than 1 minute after your last question according to your computer.

google using
DID documentation site:lists.digium.com
or
http://www.google.com/search?hl=en&ie=UTF-8&oe=UTF-8&q=DID+documentation+site%3Alists.digium.com&btnG=Google+Search

5 lines down you will find
http://lists.digium.com/pipermail/asterisk-users/2003-January/007077.html

[Paul Crick]

It's pretty easy, in your extensions.conf.

If your DIDs are in a range, you could set up some pattern matching to take
a block of incoming DIDs and map to extension numbers then dial or hand off
to the dial'n'voicemail macro thing. If your DIDs are non-contiguous, you'll
have to set up a separate entry for each one.

Example: 456-7000 is your main number and you have 7001 to 7099 as DIDs:

exten => 7000,1,Goto(AutoAttendant|s|1)
exten => _7XXX,1,Macro(yourdialmacro|${EXTEN})

[Babak Pasdar]

Hello,

I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference.

When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible.

Thanks

Babak

[Brian West]

Works fine on my 7960 with 5.3 firmware.

bkw

Post by Babak Pasdar
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference.
When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible.
Thanks
Babak
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[Juan J. Sierralta P.]

Post by Brian West
Works fine on my 7960 with 5.3 firmware.
bkw

I´m having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.

Post by Brian West

Post by Babak Pasdar
Hello,
I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference.
When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible.
Thanks
Babak

[Brian West]

=09I=C2=B4m having a similar problem with my 7960 when I receive two inco=

ming

calls I cannot join them.

ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!

bkw

[Juan J. Sierralta P.]

Post by Brian West

Post by Juan J. Sierralta P.
I´m having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.

ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!

How ? I don´t know how to park a call with the 7960.
BTW, here´s an URL which may be related with the 3-way bug on 7960s.
http://paf.se/inoc-dba/17.html

[Brian West]

I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.

bkw

=09I=C2=B4m having a similar problem with my 7960 when I receive two =

incoming

calls I cannot join them.

ya you can't join them. That sucks.. but you can park one call, go ba=

ck

to call number 1. Press conf. Dial the parking orbit.. then press joi=

n!

=09How ? I don=C2=B4t know how to park a call with the 7960.
=09BTW, here=C2=B4s an URL which may be related with the 3-way bug on 796=

0s.

=09http://paf.se/inoc-dba/17.html
--
Juanjo sin .sig
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[Babak Pasdar]

Brian,

Would you be kind enough to give me a brief overview of why it doesnt work. I also appreciate the work aorund. This is something I will have to educate my soon to be users on. We do a lot of conferencing of calls as a matter of facilitating clients' immediate needs.

For now I will try parking one or more of the calls and conferencing via calling the park extension.

I already have a meetme room setup, but it's not quite as convenient as asking someone to hangon while you get the other parties on the line to work out an issue. Especially since it is our policy to authenticate all meetmes.

Thanks for everyone's response to this issue.

Babak

Post by Brian West
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.
bkw

Post by Juan J. Sierralta P.

Post by Brian West

Post by Juan J. Sierralta P.
I´m having a similar problem with my 7960 when I receive two incoming
calls I cannot join them.

ya you can't join them. That sucks.. but you can park one call, go back
to call number 1. Press conf. Dial the parking orbit.. then press join!

How ? I don´t know how to park a call with the 7960.
BTW, here´s an URL which may be related with the 3-way bug on 7960s.
http://paf.se/inoc-dba/17.html
--
Juanjo sin .sig
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[Juan J. Sierralta P.]

Post by Babak Pasdar
Brian,
Would you be kind enough to give me a brief overview of why it doesnt work. I also appreciate the work aorund. This is something I will have to educate my soon to be users on. We do a lot of conferencing of calls as a matter of facilitating clients' immediate needs.

I still cannot park calls on my 7960, I have:

----- extensions.conf -------
[demo]
; Juanjo
exten => 8991,1,Dial(SIP/8991,20)|t
exten => 8991,2,Voicemail2(***@demo)
exten => 8991,102,Voicemail2(***@demo)
exten => 8991,103,Hangup

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => cell
include => iaxtel700
include => trunktollfree
include => iaxprovider

------ parking.conf -----------

[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in

----- sip.conf ----------------
[8991]
type=friend
username=8991
secret=secret
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=500 ; Qualify peer is no more than 200ms
context=local
mailbox=***@demo



If I dial 700 I got busy tone (440 Not Found) the same happens if I
dial #700 which I had to configure in dialplan.xml of the phone
(rewriting 700 as #700).

Any suggestions ?

[Juan J. Sierralta P.]

Post by Brian West
I dont see it as a bug.. I see why it don't work.. and why people think it
should. Enable # transfers.. and setup call parking to get around this.
Also if you conf very much look at app_meetme.

I did it, problem that I have now is the dialplan on the Cisco phone,
as soon as I push # it dial without any number :(
I´m trying to get some info on dialplan.xml if somebody has an example
to avoid the effect of the # I will appreciate it.

Thanks!

[Andrew Kohlsmith]

Post by Paul Crick
exten => 7000,1,Goto(AutoAttendant|s|1)
exten => _7XXX,1,Macro(yourdialmacro|${EXTEN})

How are you dropping the 456 there? I thought extensions picked up what
either the SIP phone had dialled, or what DTMF detection picked up when *
answered the line...?

I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer from Bell
Canada) and routing the calls coming in to multiple remote * boxes based on
the called number. I was going to ask a question similar to John's but
just didn't get around to it yet. :-)

If you could explain in a little more detail how you turn the CNID into an
extension I'd really appreciate it.

Andrew

[Paul Liew]

The number of digits that your telco sends to you is a configurable figure
(at least it is here in Aus). The example assumes that the telco is sending
you the last 4 digits.

Paul

Post by Andrew Kohlsmith

Post by Paul Crick
exten => 7000,1,Goto(AutoAttendant|s|1)
exten => _7XXX,1,Macro(yourdialmacro|${EXTEN})

How are you dropping the 456 there? I thought extensions picked up what
either the SIP phone had dialled, or what DTMF detection picked up when *
answered the line...?
I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer from Bell
Canada) and routing the calls coming in to multiple remote * boxes based on
the called number. I was going to ask a question similar to John's but
just didn't get around to it yet. :-)
If you could explain in a little more detail how you turn the CNID into an
extension I'd really appreciate it.
Andrew
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[Andrew Kohlsmith]

Post by Paul Liew
The number of digits that your telco sends to you is a configurable
figure (at least it is here in Aus). The example assumes that the telco
is sending you the last 4 digits.

Hmmm ok so DIDs are not what is stuffed into the CNID field? Or rather
pieces of the DID make it into the CNID?

Confused,
Andrew

[Steven Critchfield]

Post by Andrew Kohlsmith

Post by Paul Liew
The number of digits that your telco sends to you is a configurable
figure (at least it is here in Aus). The example assumes that the telco
is sending you the last 4 digits.

Hmmm ok so DIDs are not what is stuffed into the CNID field? Or rather
pieces of the DID make it into the CNID?

The telco will send x(configurable) digits as CNID on a PRI. This
becomes the extension inside of asterisk automagically. I know you can
have anywhere from 3 digits up to the entire 10 digits sent to you.
While my company was on channelized e&m t1 we had 4 digits of DID, and
now on PRI we get the whole 10 digit number.

[Paul Crick]

There's been a few replies but thought I'd elaborate on my initial reply..

Post by Andrew Kohlsmith
How are you dropping the 456 there? I thought extensions picked
up what either the SIP phone had dialled, or what DTMF detection
picked up when * answered the line...?

No.. if you have a PRI, the signalling is digital, no DTMFs there.. so
Asterisk received the caller ID and dialed number as part of the call setup
message. I should have explained in my example that I was assuming your
telco was sending you 4 digit DNIS. The stuff I used to work on previously,
we'd always ask for full 10 digit DNIS. Easier that way, you know exactly
what's going on (and no possibility of clashes if you have DIDs from
different exchanges).

Post by Andrew Kohlsmith
I'm looking at purchasing a PRI with 30 DIDs (can't get any fewer
from Bell Canada)

Out of curiosity, where are you located and what's the PRI cost? (I'm in
Vancouver and looking to get a T1 in the very near future)

Post by Andrew Kohlsmith
and routing the calls coming in to multiple remote * boxes based
on the called number.

So a sort of central hub/switch, taking calls in then farming them out to
remote * boxes over IP?

Cheers
Paul

[Andrew Joakimsen]

How are you transfering to 700? You dial # while in a call and then it
says "transfer" and you then dial 700, or are you using a different
method?

-----Original Message-----
Sent: Tuesday, October 07, 2003 6:02 PM
Subject: [Asterisk-Users] Call park on SIP phones

Post by Babak Pasdar
Brian,
Would you be kind enough to give me a brief overview of why it

doesnt

work. I also appreciate the work aorund. This is something I will

have

to educate my soon to be users on. We do a lot of conferencing of

calls

as a matter of facilitating clients' immediate needs.
----- extensions.conf -------
[demo]
; Juanjo
exten => 8991,1,Dial(SIP/8991,20)|t
exten => 8991,103,Hangup
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => cell
include => iaxtel700
include => trunktollfree
include => iaxprovider
------ parking.conf -----------
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
----- sip.conf ----------------
[8991]
type=friend
username=8991
secret=secret
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=500 ; Qualify peer is no more than 200ms
context=local
If I dial 700 I got busy tone (440 Not Found) the same happens

if I

dial #700 which I had to configure in dialplan.xml of the phone
(rewriting 700 as #700).
Any suggestions ?
--
Juanjo sin .sig
_______________________________________________
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Juan J. Sierralta P.]

Post by Andrew Joakimsen
How are you transfering to 700? You dial # while in a call and then it
says "transfer" and you then dial 700, or are you using a different
method?

If I dial # while in a call nothing happens. I was transfering using
the 7960 transfer function which gives me a dial tone and then I dial
700 which gives me a busy tone I also tried to dial #700 but as soon as
you push # on a 7960 it dials since # its used to signal the end of the
dial string.

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- extensions.conf -------
[demo]
; Juanjo
exten => 8991,1,Dial(SIP/8991,20)|t
exten => 8991,103,Hangup
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => cell
include => iaxtel700
include => trunktollfree
include => iaxprovider
------ parking.conf -----------
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
----- sip.conf ----------------
[8991]
type=friend
username=8991
secret=secret
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
qualify=500 ; Qualify peer is no more than 200ms
context=local
If I dial 700 I got busy tone (440 Not Found) the same happens

if I

Post by Juan J. Sierralta P.
dial #700 which I had to configure in dialplan.xml of the phone
(rewriting 700 as #700).
Any suggestions ?
--
Juanjo sin .sig
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[Brian West]

Post by Juan J. Sierralta P.
If I dial # while in a call nothing happens. I was transfering using
the 7960 transfer function which gives me a dial tone and then I dial
700 which gives me a busy tone I also tried to dial #700 but as soon as
you push # on a 7960 it dials since # its used to signal the end of the
dial string.

Using the transfer key is what I ment when I wanted native sip to parking
transfers to work. They do not now. But you have to enable T or t on
your dial command that rings your extension so # transfers will work.

bkw

[Andrew Joakimsen]

You need to enable transfer:

Dial
Dialing Application - Place an call and connect to the current channel
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][
|URL]): Requests one or more channels and places specified outgoing
calls on them. As soon as a channel answers, the Dial app will answer
the originating channel (if it needs to be answered) and will bridge a
call with the channel which first answered. All other calls placed by
the Dial app will be hunp up If a timeout is not specified, the Dial
application will wait indefinitely until either one of the called
channels answers, the user hangs up, or all channels return busy or
error. In general, the dialler will return 0 if it was unable to place
the call, or the timeout expired. However, if all channels were busy,
and there exists an extension with priority n+101 (where n is the
priority of the dialler instance), then it will be the next executed
extension (this allows you to setup different behavior on busy from
no-answer). This application returns -1 if the originating channel hangs
up, or if the call is bridged and either of the parties in the bridge
terminate the call. The option string may contain zero or more of the
following characters:
***'t' -- allow the called user transfer the calling user*** OR

***'T' -- to allow the calling user to transfer the call.***

'r' -- indicate ringing to the calling party, pass no audio until
answered.

'm' -- provide hold music to the calling party until answered.

'd' -- data-quality (modem) call (minimum delay).

'c' -- clear-channel data call (PRI-PRI only).

'H' -- allow caller to hang up by hitting *.

'C' -- reset call detail record for this call.

'P[(x)]' -- privacy mode, using 'x' as database if provided.
In addition to transferring the call, a call may be parked and then
picked up by another user. The optionnal URL will be sent to the called
party if the channel supports it.

-----Original Message-----
Sent: Tuesday, October 07, 2003 6:46 PM
Subject: RE: [Asterisk-Users] Call park on SIP phones

Post by Andrew Joakimsen
How are you transfering to 700? You dial # while in a call and then

it

Post by Andrew Joakimsen
says "transfer" and you then dial 700, or are you using a different
method?

If I dial # while in a call nothing happens. I was transfering

using

the 7960 transfer function which gives me a dial tone and then I dial
700 which gives me a busy tone I also tried to dial #700 but as soon

as

you push # on a 7960 it dials since # its used to signal the end of

the

dial string.

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- extensions.conf -------
[demo]
; Juanjo
exten => 8991,1,Dial(SIP/8991,20)|t
exten => 8991,103,Hangup
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => cell
include => iaxtel700
include => trunktollfree
include => iaxprovider
------ parking.conf -----------
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are

in

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- sip.conf ----------------
[8991]
type=friend
username=8991
secret=secret
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite

sometimes

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
qualify=500 ; Qualify peer is no more than

200ms

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
context=local
If I dial 700 I got busy tone (440 Not Found) the same happens

if I

Post by Juan J. Sierralta P.
dial #700 which I had to configure in dialplan.xml of the phone
(rewriting 700 as #700).
Any suggestions ?
--
Juanjo sin .sig
_______________________________________________
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users

--
Juanjo sin .sig
_______________________________________________
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Juan J. Sierralta P.]

I did, the problem I was using "|" instead of "," as the separator for
the Dial command, using "," solved the problem.
No idea why the syntax on the help commands shows "|" instead of ",".

[Brian West]

Yes but you can't do native sip tranfers to parking. Thats what I want.
And thats what I was talking about. You can't say use a Cisco 7960 and
hit transfer then dial 700 then transfer. WONT WORK.

bkw

Post by Andrew Joakimsen
Dial
Dialing Application - Place an call and connect to the current channel
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][
|URL]): Requests one or more channels and places specified outgoing
calls on them. As soon as a channel answers, the Dial app will answer
the originating channel (if it needs to be answered) and will bridge a
call with the channel which first answered. All other calls placed by
the Dial app will be hunp up If a timeout is not specified, the Dial
application will wait indefinitely until either one of the called
channels answers, the user hangs up, or all channels return busy or
error. In general, the dialler will return 0 if it was unable to place
the call, or the timeout expired. However, if all channels were busy,
and there exists an extension with priority n+101 (where n is the
priority of the dialler instance), then it will be the next executed
extension (this allows you to setup different behavior on busy from
no-answer). This application returns -1 if the originating channel hangs
up, or if the call is bridged and either of the parties in the bridge
terminate the call. The option string may contain zero or more of the
***'t' -- allow the called user transfer the calling user*** OR
***'T' -- to allow the calling user to transfer the call.***
'r' -- indicate ringing to the calling party, pass no audio until
answered.
'm' -- provide hold music to the calling party until answered.
'd' -- data-quality (modem) call (minimum delay).
'c' -- clear-channel data call (PRI-PRI only).
'H' -- allow caller to hang up by hitting *.
'C' -- reset call detail record for this call.
'P[(x)]' -- privacy mode, using 'x' as database if provided.
In addition to transferring the call, a call may be parked and then
picked up by another user. The optionnal URL will be sent to the called
party if the channel supports it.

-----Original Message-----
Sent: Tuesday, October 07, 2003 6:46 PM
Subject: RE: [Asterisk-Users] Call park on SIP phones

Post by Andrew Joakimsen
How are you transfering to 700? You dial # while in a call and then

it

Post by Andrew Joakimsen
says "transfer" and you then dial 700, or are you using a different
method?

If I dial # while in a call nothing happens. I was transfering

using

the 7960 transfer function which gives me a dial tone and then I dial
700 which gives me a busy tone I also tried to dial #700 but as soon

as

you push # on a 7960 it dials since # its used to signal the end of

the

dial string.

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- extensions.conf -------
[demo]
; Juanjo
exten => 8991,1,Dial(SIP/8991,20)|t
exten => 8991,103,Hangup
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => cell
include => iaxtel700
include => trunktollfree
include => iaxprovider
------ parking.conf -----------
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are

in

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- sip.conf ----------------
[8991]
type=friend
username=8991
secret=secret
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite

sometimes

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
qualify=500 ; Qualify peer is no more than

200ms

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
context=local
If I dial 700 I got busy tone (440 Not Found) the same happens

if I

Post by Juan J. Sierralta P.
dial #700 which I had to configure in dialplan.xml of the phone
(rewriting 700 as #700).
Any suggestions ?
--
Juanjo sin .sig
_______________________________________________
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
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http://lists.digium.com/mailman/listinfo/asterisk-users

--
Juanjo sin .sig
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[Shaun Ewing]

Post by Brian West
Yes but you can't do native sip tranfers to parking. Thats what I want.
And thats what I was talking about. You can't say use a Cisco 7960 and
hit transfer then dial 700 then transfer. WONT WORK.
bkw

I've implemented a bit of a workaround.

I've setup the dial plan 2XXXX in my system as the call park prefix.

When you want to park a call, you blind transfer to 2XXXX where XXXX is your
extension (eg: 27011). The call is parked, and you will immediately receive
a call announcing the park slot (eg: "Transfer 8 0 0 1").

The appropriate entry in extensions.conf looks like:

[Shaun Ewing]

Apologies for the previous message, I accidentally hit send prematurely.

Post by Brian West
Yes but you can't do native sip tranfers to parking. Thats what I want.
And thats what I was talking about. You can't say use a Cisco 7960 and
hit transfer then dial 700 then transfer. WONT WORK.
bkw

I've implemented a bit of a workaround.

I've setup the dial plan 2XXXX in my system as the call park prefix.

When you want to park a call, you blind transfer to 2XXXX where XXXX is your
extension (eg: 27011). The call is parked, and you will immediately receive
a call announcing the park slot (eg: "Transfer 8 0 0 1").

The appropriate entry in extensions.conf looks like:
exten => _2XXXX,1,Answer
exten =>
_2XXXX,2,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|local-exten
sions,${EXTEN:1},1)

This only works if the extension is SIP, but I'm sure it would be possible
to modify for others.

-Shaun

[Brian West]

Post by Shaun Ewing
I've implemented a bit of a workaround.
I've setup the dial plan 2XXXX in my system as the call park prefix.
When you want to park a call, you blind transfer to 2XXXX where XXXX is your
extension (eg: 27011). The call is parked, and you will immediately receive
a call announcing the park slot (eg: "Transfer 8 0 0 1").
exten => _2XXXX,1,Answer
exten =>
_2XXXX,2,ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|local-exten
sions,${EXTEN:1},1)

Nice work around.... :P

[Andrew Joakimsen]

Because otherwise the parked extension will not be announced.

-----Original Message-----
Sent: Tuesday, October 07, 2003 9:03 PM
Subject: RE: [Asterisk-Users] Call park on SIP phones
Yes but you can't do native sip tranfers to parking. Thats what I

want.

And thats what I was talking about. You can't say use a Cisco 7960

and

hit transfer then dial 700 then transfer. WONT WORK.
bkw

Post by Andrew Joakimsen
Dial
Dialing Application - Place an call and connect to the current

channel
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][

Post by Andrew Joakimsen
|URL]): Requests one or more channels and places specified outgoing
calls on them. As soon as a channel answers, the Dial app will

answer

Post by Andrew Joakimsen
the originating channel (if it needs to be answered) and will bridge

a

Post by Andrew Joakimsen
call with the channel which first answered. All other calls placed

by

Post by Andrew Joakimsen
the Dial app will be hunp up If a timeout is not specified, the Dial
application will wait indefinitely until either one of the called
channels answers, the user hangs up, or all channels return busy or
error. In general, the dialler will return 0 if it was unable to

place

Post by Andrew Joakimsen
the call, or the timeout expired. However, if all channels were

busy,

Post by Andrew Joakimsen
and there exists an extension with priority n+101 (where n is the
priority of the dialler instance), then it will be the next executed
extension (this allows you to setup different behavior on busy from
no-answer). This application returns -1 if the originating channel

hangs

Post by Andrew Joakimsen
up, or if the call is bridged and either of the parties in the

bridge

Post by Andrew Joakimsen
terminate the call. The option string may contain zero or more of

the

Post by Andrew Joakimsen
***'t' -- allow the called user transfer the calling user*** OR
***'T' -- to allow the calling user to transfer the call.***
'r' -- indicate ringing to the calling party, pass no audio until
answered.
'm' -- provide hold music to the calling party until answered.
'd' -- data-quality (modem) call (minimum delay).
'c' -- clear-channel data call (PRI-PRI only).
'H' -- allow caller to hang up by hitting *.
'C' -- reset call detail record for this call.
'P[(x)]' -- privacy mode, using 'x' as database if provided.
In addition to transferring the call, a call may be parked and then
picked up by another user. The optionnal URL will be sent to the

called

Post by Andrew Joakimsen
party if the channel supports it.

-----Original Message-----

[mailto:asterisk-users-

Post by Andrew Joakimsen

Sent: Tuesday, October 07, 2003 6:46 PM
Subject: RE: [Asterisk-Users] Call park on SIP phones

Post by Andrew Joakimsen
How are you transfering to 700? You dial # while in a call and

then

Post by Andrew Joakimsen
it

Post by Andrew Joakimsen
says "transfer" and you then dial 700, or are you using a

different

Post by Andrew Joakimsen

Post by Andrew Joakimsen
method?

If I dial # while in a call nothing happens. I was transfering

using

the 7960 transfer function which gives me a dial tone and then I

dial

Post by Andrew Joakimsen

700 which gives me a busy tone I also tried to dial #700 but as

soon

Post by Andrew Joakimsen
as

you push # on a 7960 it dials since # its used to signal the end

of

Post by Andrew Joakimsen
the

dial string.

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- extensions.conf -------
[demo]
; Juanjo
exten => 8991,1,Dial(SIP/8991,20)|t
exten => 8991,103,Hangup
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => cell
include => iaxtel700
include => trunktollfree
include => iaxprovider
------ parking.conf -----------
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls

are

Post by Andrew Joakimsen
in

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
----- sip.conf ----------------
[8991]
type=friend
username=8991
secret=secret
nat=no ; This phone may be natted
host=dynamic
canreinvite=no ; Cisco poops on reinvite

sometimes

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
qualify=500 ; Qualify peer is no more than

200ms

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
context=local
If I dial 700 I got busy tone (440 Not Found) the same

happens

Post by Andrew Joakimsen

Post by Andrew Joakimsen
if I

Post by Juan J. Sierralta P.
dial #700 which I had to configure in dialplan.xml of the

phone

Post by Andrew Joakimsen

Post by Andrew Joakimsen

Post by Juan J. Sierralta P.
(rewriting 700 as #700).
Any suggestions ?
--
Juanjo sin .sig
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--
Juanjo sin .sig
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[Nicolas Gudino]

Hi List,

I'm new to asterisk. I think it's great! I'm interested in terminating calls
via a SIP provider. I want to know if I need to license G729 on asterisk in
these scenarios:

CISCO ATA186 - Asterisk - SIP Provider - PSTN

or this one:

CISCO ATA186 - Asterisk - CISCO ATA

To my understanding, in the second case, if one of the ATA is behind NAT, I
should set canreinvite=no, so the RTP channels would go through *, so I
would have to license G729 in order to use this codec with the ATAs. Is this
right?

But if boths ATA have public IPs, and * issues a reinvite, can the ATAs
negotiate G729 themselves, without needing it on * ?

And in the first scenario, if the SIP provider supports G729 and the ATA has
a public IP, do I need to license the codec in *?

Thanks in advance,

Nicolas Gudino
Buenos Aires - Argentina

[Eric Wieling]

Since Asterisk isn't converting from one codec to another you should not
need a G.729 license.

Post by Nicolas Gudino
Hi List,
I'm new to asterisk. I think it's great! I'm interested in terminating calls
via a SIP provider. I want to know if I need to license G729 on asterisk in
CISCO ATA186 - Asterisk - SIP Provider - PSTN
CISCO ATA186 - Asterisk - CISCO ATA
To my understanding, in the second case, if one of the ATA is behind NAT, I
should set canreinvite=no, so the RTP channels would go through *, so I
would have to license G729 in order to use this codec with the ATAs. Is this
right?
But if boths ATA have public IPs, and * issues a reinvite, can the ATAs
negotiate G729 themselves, without needing it on * ?
And in the first scenario, if the SIP provider supports G729 and the ATA has
a public IP, do I need to license the codec in *?
Thanks in advance,
Nicolas Gudino
Buenos Aires - Argentina
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