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[asterisk-users] Attended transfers manager or phone


 
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chr.ejlertsen at has.dk
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PostPosted: Tue Jan 15, 2008 3:32 pm    Post subject: [asterisk-users] Attended transfers manager or phone Reply with quote

Well I'm sure this issue has been bean up a few time since it's one of the
only ones I can't find a real "simple" answer to.

I'm trying to find away to do attended transfers through the manager
interface, for a pc switchboard / Agent client solution, but so far coming
up short.
The action Originate is part of the solution, but what really I want is the
phone being taken off-hook and then being able to dial the number without
having to answer the dial-back first.

1. One solution, though an ugly one, would be using Originate, but use a
phone that has some sort tcp/ip interface that allows for taking the phone
off-hook.

2. A Better solution would be using a phone that allows dialling and taking
the phone off-hook on-hook etc. via some tcp/ip interface.

3. Yet another solution, though I do not favour this one since I really
don't want to maintain the sip phone code, would be programming a soft sip
phone with all the bells and whistles and adding the switchboard
functionality to that (name searching, status email so on and so forth.

In the end all I need is just a software or hardware phone, sip/iax, which
can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps
status requests. If such a phone exists that would do the trick, the rest is
manageable via the Asterisk Manager console.

I'm guessing some people have messed with this problem before so I hope that
someone has some information about this kind of thing Smile

Thank you in advance
Christian
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PostPosted: Tue Jan 15, 2008 7:06 pm    Post subject: [asterisk-users] Attended transfers manager or phone Reply with quote

Some phones have the auto-answer ability. So your phone could have two
extensions, one for normal use and one for auto-answer use. Redirect or
Originate, as you were, to the auto-answer extension on the phone. So
the phone would already put itself offhook, and asterisk would continue
and build up the other end of the bridge.

Polycom soundpoint phones, for example, but many others have this ability.

an example extension setup might be

exten => 110,1,Dial(SIP/110)

exten => #110,1,SipAddHeader(.......whatever your phone needs to make it
autoanswer)
exten => #110,2,Dial(SIP/110)

Don't know about phones that allow ip control of their state, though.

Moj

Christian Ejlertsen wrote:
Quote:
Well I'm sure this issue has been bean up a few time since it's one of the
only ones I can't find a real "simple" answer to.

I'm trying to find away to do attended transfers through the manager
interface, for a pc switchboard / Agent client solution, but so far coming
up short.
The action Originate is part of the solution, but what really I want is the
phone being taken off-hook and then being able to dial the number without
having to answer the dial-back first.

1. One solution, though an ugly one, would be using Originate, but use a
phone that has some sort tcp/ip interface that allows for taking the phone
off-hook.

2. A Better solution would be using a phone that allows dialling and taking
the phone off-hook on-hook etc. via some tcp/ip interface.

3. Yet another solution, though I do not favour this one since I really
don't want to maintain the sip phone code, would be programming a soft sip
phone with all the bells and whistles and adding the switchboard
functionality to that (name searching, status email so on and so forth.

In the end all I need is just a software or hardware phone, sip/iax, which
can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps
status requests. If such a phone exists that would do the trick, the rest is
manageable via the Asterisk Manager console.

I'm guessing some people have messed with this problem before so I hope that
someone has some information about this kind of thing Smile

Thank you in advance
Christian


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chr.ejlertsen at has.dk
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PostPosted: Wed Jan 16, 2008 10:24 am    Post subject: [asterisk-users] Attended transfers manager or phone Reply with quote

Thank you very much, that was a new angle I hadn't thought of time to
investigate a little more Smile. The joys of learning new things Smile

- Christian

Quote:
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
bounces at lists.digium.com] On Behalf Of Mojo with Horan & Company, LLC
Sent: 16. januar 2008 01:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Attended transfers manager or phone

Some phones have the auto-answer ability. So your phone could have two
extensions, one for normal use and one for auto-answer use. Redirect or
Originate, as you were, to the auto-answer extension on the phone. So
the phone would already put itself offhook, and asterisk would continue
and build up the other end of the bridge.

Polycom soundpoint phones, for example, but many others have this ability.

an example extension setup might be

exten => 110,1,Dial(SIP/110)

exten => #110,1,SipAddHeader(.......whatever your phone needs to make it
autoanswer)
exten => #110,2,Dial(SIP/110)

Don't know about phones that allow ip control of their state, though.

Moj

Christian Ejlertsen wrote:
Quote:
Well I'm sure this issue has been bean up a few time since it's one of
the
Quote:
only ones I can't find a real "simple" answer to.

I'm trying to find away to do attended transfers through the manager
interface, for a pc switchboard / Agent client solution, but so far
coming
Quote:
up short.
The action Originate is part of the solution, but what really I want is
the
Quote:
phone being taken off-hook and then being able to dial the number
without
Quote:
having to answer the dial-back first.

1. One solution, though an ugly one, would be using Originate, but use a
phone that has some sort tcp/ip interface that allows for taking the
phone
Quote:
off-hook.

2. A Better solution would be using a phone that allows dialling and
taking
Quote:
the phone off-hook on-hook etc. via some tcp/ip interface.

3. Yet another solution, though I do not favour this one since I really
don't want to maintain the sip phone code, would be programming a soft
sip
Quote:
phone with all the bells and whistles and adding the switchboard
functionality to that (name searching, status email so on and so forth.

In the end all I need is just a software or hardware phone, sip/iax,
which
Quote:
can be told via tcp/ip to go off-hook, on-hook, dial, transfer and
perhaps
Quote:
status requests. If such a phone exists that would do the trick, the
rest is
Quote:
manageable via the Asterisk Manager console.

I'm guessing some people have messed with this problem before so I hope
that
Quote:
someone has some information about this kind of thing Smile

Thank you in advance
Christian


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_______________________________________________
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