Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Mueller, Alexander
OK, as promised, here are the details of the solution.

The local phone which shall dial out is SIP/2000, and the external number to be 
called is 0123456789. The SIP/2000 will auto-answer, so that the user (having a 
headset on) will be able to make an outbound call without using the phone 
hardware.

The Action to be called from the AMI is:

Action: Originate
Channel: Local/*2...@originating
Context: originating
Exten: 00123456789
Priority: 1
CallerID: 2000
ActionID: ORIGINATE_464

The "*" is only a marker for the distinction inside the dialplan, to 
distinguish internal and external phone numbers.
The dialplan goes:

<...>
; Originating Calls
[originating]
exten => _0X.,1,Dial(SIP/${EXTEN:1...@sipgate-out)

exten => _*X.,1,Set(NST=${EXTEN:1})
exten => _*X.,n,SIPAddHeader(Call-Info: sip:\;answer-after=0)
exten => _*X.,n,Dial(SIP/${NST})
<...>

Where you can see the "*" to distinguish internal and external. I use 
"sipgate-out" as context for outbound calls.

This solution has no checking if SIP/2000 is busy or not, but when a user who 
"owns" the phone presses the button (and triggers the originate command), I 
think he knows why something went wrong if the phone is not ready.

Thanks everybody for the hints !

Alex

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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Mueller, Alexander
Thanks, I found the solution using the local/ way.

I didn't find it documented in the Asterisk book, whether the English one (van 
Meggelen), nor the german one (Wintermeyer/Kempgen).

But this link brought me to the solution: 
http://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.html
 

Thanks to everyone helping me.

Later this day I will document the solution in detail here for others having 
the same problem

Alex



-Ursprüngliche Nachricht-
Von: asterisk-biz-boun...@lists.digium.com 
[mailto:asterisk-biz-boun...@lists.digium.com] Im Auftrag von Olle E. Johansson
Gesendet: Sonntag, 29. November 2009 19:43
An: Commercial and Business-Oriented Asterisk Discussion
Betreff: Re: [asterisk-biz] auto-answering an originated call, dialplan / 
manager interface problem


29 nov 2009 kl. 17.47 skrev Mueller, Alexander:

> 29.11.  17:37:19,060 Action: Originate
> Channel: SIP/2000

> Context: originating
> Exten: #*00123456798
> Priority: 1
> CallerID: 2000
> Variable: Outbound_CALLERID=07615987654321
> ActionID: ORIGINATE_452


Instead of SIP/2000 you can define local/ex...@context and in that extension 
add SIP headers and perform any magic you want.

/O


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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Olle E. Johansson

29 nov 2009 kl. 17.47 skrev Mueller, Alexander:

> 29.11.  17:37:19,060 Action: Originate
> Channel: SIP/2000

> Context: originating
> Exten: #*00123456798
> Priority: 1
> CallerID: 2000
> Variable: Outbound_CALLERID=07615987654321
> ActionID: ORIGINATE_452


Instead of SIP/2000 you can define local/ex...@context and in that extension 
add SIP headers and perform any magic you want.

/O


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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Olle E. Johansson

29 nov 2009 kl. 15.32 skrev Kevin P. Fleming:

> Olle E. Johansson wrote:
> 
>> Secondly, this mailing list is for Asterisk development - new code and 
>> issues with the Asterisk code. Your questions would get a faster and better 
>> answer on asterisk-users. Even if it's about development, it's about 
>> development of your own app, not about Asterisk.
> 
> Actually, this is the -biz list :-)
> 
Oh, one of those days. Then it's much much better to ask for advice on the 
asterisk-users list!

Sorry.

/Olle
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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Mueller, Alexander
CS, thanks, both this ways I already went, but they don't solve the problem.

The originate way originates my call, but the agent has to press the 
acknowledge button on his phone, which shall not occur (the call shall be 
established without the agent needing to use the hardware phone). The dialplan 
is entered much too late by using this way, it's entered *after* the user has 
pressed a key on the phone (cancel or acknowledge).

Here's the event log coming from Asterisk (times inserted by my logging app).
At 17:37:19, the phone starts ringing, without the dialplan even being reached 
(SIPAddHeader!)
At 17:37:32, I acknowledge on the phone, then come into the dialplan, and can 
call SIPAddHeader, that's too late.


29.11.  17:37:19,060 Action: Originate
Channel: SIP/2000
Context: originating
Exten: #*00123456798
Priority: 1
CallerID: 2000
Variable: Outbound_CALLERID=07615987654321
ActionID: ORIGINATE_452


29.11.  17:37:32,660 Response: Success
ActionID: ORIGINATE_452
Message: Originate successfully queued

Event: Newchannel
Privilege: call,all
Channel: SIP/2000-08bba8b8
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum: 
CallerIDName: 
AccountCode: 
Uniqueid: 1259512663.0

<...>

Event: Newexten
Privilege: dialplan,all
Channel: SIP/2000-08bba8b8
Context: originating
Extension: #*00123456798
Priority: 2
Application: SIPAddHeader
AppData: Call-Info: sip:\;answer-after=0
Uniqueid: 1259512663.0

<...>

Event: Newexten
Privilege: dialplan,all
Channel: SIP/2000-08bba8b8
Context: originating
Extension: 00123456798
Priority: 3
Application: Dial
AppData: 00123456798
Uniqueid: 1259512663.0




-Ursprüngliche Nachricht-
Von: asterisk-biz-boun...@lists.digium.com 
[mailto:asterisk-biz-boun...@lists.digium.com] Im Auftrag von 
c.savinov...@itntelecom.com
Gesendet: Sonntag, 29. November 2009 15:32
An: asterisk-biz@lists.digium.com
Betreff: Re: [asterisk-biz] auto-answering an originated call, dialplan / 
manager interface problem


Alexander:

Once you get the handle of it, it ain't difficult:


AMI Example:

Action: Originate
Channel: ---> this is leg 1, the extension/channel "making" the call.
(actually, asterisk will call this extension /channel first)
Context: ---> Here is the context name in your dialplan(leg 2), for
example, "mycontext"
Exten:   ---> Exten, example "myexten"

Dialplan Example:

[mycontext]
myexten, 1, SipAddHeader(headertext)
myexten, 2, Dial(dialstring)


C. Savinovich



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Re: [asterisk-biz] Global Crossing

2009-11-29 Thread Jared Geiger
Ah yes, we're on Gold rates. Time to call and inquire about Platinum rates
:)

On Sun, Nov 29, 2009 at 4:36 AM, Yaro Donchenko
wrote:

> I think you had a wrong rates because you did not get a Platinum
> account. But $250k sounds right deposit ;-)
> If you looking to get a interconnection with Tier 1 carries and save
> some money on down payments please contact me of the list.
>
> Regards
> Yaro.
>
> Jared Geiger wrote:
> > I haven't had any problems with Qwest's wholesale service itself. Only
> > with the customer service have I run into issues. Heh my deposit was
> > much lower than anything quoted here too. Granted the rates aren't
> > exactly stellar now though.
> >
> > On Mon, Nov 23, 2009 at 7:19 PM, Wes Reece  > > wrote:
> >
> > Yes we have qwest, we put down $250k and it was a complete waste.
> > we tied up cash for sh*ty lines.
> >
> >
> >
> > On Sun, Nov 22, 2009 at 3:56 PM, Yaro Donchenko
> > mailto:ydonche...@logicvoice.com>>
> wrote:
> >
> > $40k is not a large deposit. Try Qwest you will see what is
> > large deposit.
> > --
> > Best Regards,
> > Yaro Donchenko
> > Logic Voice, Inc.
> > Direct: (718) 928-6700
> > Main: (718) 928-
> > Direct Fax: (718) 569-6900
> > Fax: (718) 928-6677
> > URL: www.logicvoice.com 
> >
> > ICQ: 825979
> > AIM: ydonche...@logicvoice.com  >
> > MSN: ydonche...@logicvoice.com  >
> > Jabber: ydonche...@logicvoice.com
> > 
> >
> >
> > On Thu, Nov 19, 2009 at 5:43 AM, Rehan Allah Wala
> > mailto:re...@supertec.com>> wrote:
> >
> > So are you up for buying at 10% from me ?
> >
> >
> > if u do, kindly do email me directly
> >
> > >
> > > Dave,
> > >
> > > That was a replay to Rehan, not intended for you. And
> > you're right, most people like to keep a
> > > 30% markup which is pretty tough in the wholesale
> > minutes business. 10% I barely cover my
> > > operating expenses.
> > >
> > > On Mon, Nov 16, 2009 at 9:20 PM, David Knell
> > mailto:d...@3c.co.uk>> wrote:
> > > Hi Wes -
> > >
> > > > Yes I got 1/1 billing. No I'm not interested in buying
> > from resellers,
> > > > but thanks for the offer and 10% markup? WOW...I wish
> > I could charge
> > > > my customers that much!
> > >
> > > Not sure quite where this came from, but I don't
> > recall offering you
> > > anything. And, if you can't make a 10% gross margin
> > on stuff, you
> > > probably ought to do something else ;-)
> > >
> > > --Dave
> > >
> > >
> > > >
> > > >
> > > >
> > > > On Sun, Nov 15, 2009 at 10:21 AM, David Knell
> > mailto:d...@3c.co.uk>> wrote:
> > > > Yes. Pretty good.
> > > >
> > > > --Dave
> > > >
> > > >
> > > > > Anybody have recent experience with them?
> > Specifically
> > > > regarding
> > > > > termination on their A-Z routes? They're asking
> > for a fairly
> > > > large
> > > > > deposit ($40k), and I want to make sure I'm
> > moving in the
> > > > right
> > > > > direction before tying that money up.
> > > > >
> > > > > --
> > > > > Thank you,
> > > > >
> > > > > -Wes-
> > > >
> > > >
> > > > > ___
> > > > > --Bandwidth and Colocation Provided by
> > > > http://www.api-digital.com--
> > > > >
> > > > > asterisk-biz mailing list
> > > > > To UNSUBSCRIBE or update options visit:
> > > > >
> >  http://lists.digium.com/mailman/listinfo/asterisk-biz
> > > >
> > > >
> > > > ___
> > > > --Bandwidth and Colocation Provided by
> > > > http://www.api-digital.com--
> > > >
> > > > asterisk-biz mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
> http://lists.digium.com/mailman/listinfo/asterisk-biz
> > > >
> > > >
> > > >
> > > >
> >

Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Kevin P. Fleming
Olle E. Johansson wrote:

> Secondly, this mailing list is for Asterisk development - new code and issues 
> with the Asterisk code. Your questions would get a faster and better answer 
> on asterisk-users. Even if it's about development, it's about development of 
> your own app, not about Asterisk.

Actually, this is the -biz list :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread c . savinovich

Alexander:

Once you get the handle of it, it ain't difficult:


AMI Example:

Action: Originate
Channel: ---> this is leg 1, the extension/channel "making" the call.
(actually, asterisk will call this extension /channel first)
Context: ---> Here is the context name in your dialplan(leg 2), for
example, "mycontext"
Exten:   ---> Exten, example "myexten"

Dialplan Example:

[mycontext]
myexten, 1, SipAddHeader(headertext)
myexten, 2, Dial(dialstring)


C. Savinovich



>>>CS, that's my problem, exactly how you describe it:

>>> but why can't you redirect the AMI Originate to a dialplan script that
uses the SipAddHeader?

After lots of experimenting, I now come from AMI into the dialplan by just
calling "Action: Command\r\nCommand: console dial #*2000" from the AMI,
where #* is the flag for originating (which is handled in the dialplan),
and I'm in the dialplan, so far so good. I can set the SIPHeader there an
ring the local phone (2000) via "Dial", the local phone takes the call
automatically, fine.

But then I have only one leg, I have not yet found out how to connect this
first leg inside the dialplan with the second leg.

How can this originating be done, what has to come after Dial in order to
establish the outgoing call ?

I have read the description of "Dial()" several times now, hard to
understand, it doesn't seem to offer a 2-leg-addressing (internal number
and external number).

I also thought about creating call files in the "outgoing" directory, but
then again the problem with the missing SIPAddHeader() will lead to
failure.

Still unsolved ...



-Ursprüngliche Nachricht-
Von: asterisk-biz-boun...@lists.digium.com
[mailto:asterisk-biz-boun...@lists.digium.com] Im Auftrag von
c.savinov...@itntelecom.com
Gesendet: Samstag, 28. November 2009 03:33
An: asterisk-biz@lists.digium.com
Betreff: Re: [asterisk-biz] auto-answering an originated call, dialplan /
manager interface problem


There is nothing like SipAddHeader in the AMI, but why can't you redirect
the AMI Originate to a dialplan script that uses the SipAddHeader?

CS



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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Olle E. Johansson

29 nov 2009 kl. 12.14 skrev Mueller, Alexander:

> CS, that's my problem, exactly how you describe it:
> 
 but why can't you redirect the AMI Originate to a dialplan script that 
 uses the SipAddHeader?
> 
> After lots of experimenting, I now come from AMI into the dialplan by just 
> calling "Action: Command\r\nCommand: console dial #*2000" from the AMI, where 
> #* is the flag for originating (which is handled in the dialplan), and I'm in 
> the dialplan, so far so good. I can set the SIPHeader there an ring the local 
> phone (2000) via "Dial", the local phone takes the call automatically, fine. 
> 
> But then I have only one leg, I have not yet found out how to connect this 
> first leg inside the dialplan with the second leg.
> 
> How can this originating be done, what has to come after Dial in order to 
> establish the outgoing call ?
> 
> I have read the description of "Dial()" several times now, hard to 
> understand, it doesn't seem to offer a 2-leg-addressing (internal number and 
> external number).
> 
> I also thought about creating call files in the "outgoing" directory, but 
> then again the problem with the missing SIPAddHeader() will lead to failure.
> 
> Still unsolved ...
> 

You have to learn about the local channel and how to use that from Originate. I 
think that's covered in the Asterisk book, downloadable from asteriskdocs.org 
and in many examples on the web.

Secondly, this mailing list is for Asterisk development - new code and issues 
with the Asterisk code. Your questions would get a faster and better answer on 
asterisk-users. Even if it's about development, it's about development of your 
own app, not about Asterisk.

Good luck in your Asterisk adventures!

Regards
/Olle
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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

2009-11-29 Thread Mueller, Alexander
CS, that's my problem, exactly how you describe it:

>>> but why can't you redirect the AMI Originate to a dialplan script that uses 
>>> the SipAddHeader?

After lots of experimenting, I now come from AMI into the dialplan by just 
calling "Action: Command\r\nCommand: console dial #*2000" from the AMI, where 
#* is the flag for originating (which is handled in the dialplan), and I'm in 
the dialplan, so far so good. I can set the SIPHeader there an ring the local 
phone (2000) via "Dial", the local phone takes the call automatically, fine. 

But then I have only one leg, I have not yet found out how to connect this 
first leg inside the dialplan with the second leg.

How can this originating be done, what has to come after Dial in order to 
establish the outgoing call ?

I have read the description of "Dial()" several times now, hard to understand, 
it doesn't seem to offer a 2-leg-addressing (internal number and external 
number).

I also thought about creating call files in the "outgoing" directory, but then 
again the problem with the missing SIPAddHeader() will lead to failure.

Still unsolved ...



-Ursprüngliche Nachricht-
Von: asterisk-biz-boun...@lists.digium.com 
[mailto:asterisk-biz-boun...@lists.digium.com] Im Auftrag von 
c.savinov...@itntelecom.com
Gesendet: Samstag, 28. November 2009 03:33
An: asterisk-biz@lists.digium.com
Betreff: Re: [asterisk-biz] auto-answering an originated call, dialplan / 
manager interface problem


There is nothing like SipAddHeader in the AMI, but why can't you redirect
the AMI Originate to a dialplan script that uses the SipAddHeader?

CS



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Re: [asterisk-biz] Global Crossing

2009-11-29 Thread Yaro Donchenko
I think you had a wrong rates because you did not get a Platinum 
account. But $250k sounds right deposit ;-)
If you looking to get a interconnection with Tier 1 carries and save 
some money on down payments please contact me of the list.

Regards
Yaro.

Jared Geiger wrote:
> I haven't had any problems with Qwest's wholesale service itself. Only 
> with the customer service have I run into issues. Heh my deposit was 
> much lower than anything quoted here too. Granted the rates aren't 
> exactly stellar now though.
>
> On Mon, Nov 23, 2009 at 7:19 PM, Wes Reece  > wrote:
>
> Yes we have qwest, we put down $250k and it was a complete waste.
> we tied up cash for sh*ty lines.
>
>
>
> On Sun, Nov 22, 2009 at 3:56 PM, Yaro Donchenko
> mailto:ydonche...@logicvoice.com>> wrote:
>
> $40k is not a large deposit. Try Qwest you will see what is
> large deposit.
> --
> Best Regards,
> Yaro Donchenko
> Logic Voice, Inc.
> Direct: (718) 928-6700
> Main: (718) 928-
> Direct Fax: (718) 569-6900
> Fax: (718) 928-6677
> URL: www.logicvoice.com 
>
> ICQ: 825979
> AIM: ydonche...@logicvoice.com 
> MSN: ydonche...@logicvoice.com 
> Jabber: ydonche...@logicvoice.com
> 
>
>
> On Thu, Nov 19, 2009 at 5:43 AM, Rehan Allah Wala
> mailto:re...@supertec.com>> wrote:
>
> So are you up for buying at 10% from me ?
>
>
> if u do, kindly do email me directly
>
> >
> > Dave,
> >
> > That was a replay to Rehan, not intended for you. And
> you're right, most people like to keep a
> > 30% markup which is pretty tough in the wholesale
> minutes business. 10% I barely cover my
> > operating expenses.
> >
> > On Mon, Nov 16, 2009 at 9:20 PM, David Knell
> mailto:d...@3c.co.uk>> wrote:
> > Hi Wes -
> >
> > > Yes I got 1/1 billing. No I'm not interested in buying
> from resellers,
> > > but thanks for the offer and 10% markup? WOW...I wish
> I could charge
> > > my customers that much!
> >
> > Not sure quite where this came from, but I don't
> recall offering you
> > anything. And, if you can't make a 10% gross margin
> on stuff, you
> > probably ought to do something else ;-)
> >
> > --Dave
> >
> >
> > >
> > >
> > >
> > > On Sun, Nov 15, 2009 at 10:21 AM, David Knell
> mailto:d...@3c.co.uk>> wrote:
> > > Yes. Pretty good.
> > >
> > > --Dave
> > >
> > >
> > > > Anybody have recent experience with them?
> Specifically
> > > regarding
> > > > termination on their A-Z routes? They're asking
> for a fairly
> > > large
> > > > deposit ($40k), and I want to make sure I'm
> moving in the
> > > right
> > > > direction before tying that money up.
> > > >
> > > > --
> > > > Thank you,
> > > >
> > > > -Wes-
> > >
> > >
> > > > ___
> > > > --Bandwidth and Colocation Provided by
> > > http://www.api-digital.com--
> > > >
> > > > asterisk-biz mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
>  http://lists.digium.com/mailman/listinfo/asterisk-biz
> > >
> > >
> > > ___
> > > --Bandwidth and Colocation Provided by
> > > http://www.api-digital.com--
> > >
> > > asterisk-biz mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >  http://lists.digium.com/mailman/listinfo/asterisk-biz
> > >
> > >
> > >
> > >
> > > --
> > > Thank you,
> > >
> > > -Wes-
> > > ___
> > > --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
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> > > asterisk-biz mailing list
> > > To UNSUBSCRIBE or update options visit:
>