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[Prometheus001 at gmx.net]

I have managed to have a realtme status of a phone on a web page with
event_socket and a push service to the web bowser.

What I am now trying to do is roughly the following:

* when a call comes in, a flashing banner appears on the web page
with an underlying link (this works so far)
* when the user klicks on this flashing banner, the external SIP UA
which is already ringing, shall pick up the call.

I know that it's possible to autoanswer a call with the intercom
feature. Also the SIP client X-Lite which we use here is able to
autoanswer a call.
I however want to manually decide when the UA takes the call with the
following workflow:

* X-Lite rings on incoming call
* user klicks on the flashing banner
* X-Lite takes the call

What is the best way to have this done? Move the call to park and then
retransfer again with intercom, or is there a better solution?

Best regards
Peter



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[brian at freeswitch.org]

click on the AA button?

/b

On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:

[Prometheus001 at gmx.net]

Hello Brian,

this is too easy .

This is for a small callcenter app and I only want the user to pickup
the call once (to accept the call in X-Lite (or a Snom phone) and to
start the workflow on the web application). I do not want him to accept
the call on the phone and then on the Web app.

Best regards
Peter



Brian West schrieb:

[mike at jerris.com]

The only way I can think to do this today would be to cancel the call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it it
a sipp or sipsak scenario and test it. A better aproach might be to
answer the call normally and detect that to start your web workflow or
not really ring the phone, just the web app and deliver the call with
autoanswer when the button is hit in the web ui.

Mike

On Jun 16, 2009, at 4:24 AM, Peter P GMX <Prometheus001@gmx.net> wrote:

[intralanman at freeswi...]

Peter P GMX wrote:

[Prometheus001 at gmx.net]

Hello Ray,

I do use event socket and it pushes me a link on the website whenever a
call for this agent comes in.

It's just a matter of visibility. The agent may still finish his old
workflow and is still entering data. When a call comes in then and he
picks up the phone, the data he just entered is gone away. So I would
like the web app to drive answering the call. It gives a better
visibility about what he is doing to the callcenter agent.

Best regards
Peter

Raymond Chandler schrieb:

[Prometheus001 at gmx.net]

Hello Michael,

I want the phone be ringing, just for acoustical feedback reasons.

But what if I

* transfer it to the same user destination again (now with intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with intercom enabled)

Best regards
Peter

Michael Jerris schrieb:

[brian at freeswitch.org]

Why not just keep the agent off hook.. in park state... then just
playback ringing before you bridge?

/b

On Jun 16, 2009, at 7:38 AM, Peter P GMX wrote:

[mike at jerris.com]

The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's suggestion.



On Jun 16, 2009, at 7:38 AM, Peter P GMX <Prometheus001@gmx.net> wrote:

[raul at etellicom.com]

I actually do that with our call center application. For all incoming
calls, our IVR engine parks the call in a virtual extension and plays
back prompts, advertisements, MOH, process digits, etc. When the queue
management finds an available agent, it sends an event to the client
application for that agent (with an optional screen-pop) where the agent
can click "Answer Call" and then we transfer the call with the
auto-answer header set on to the agent phone.
You could take a similar approach, if you're worrying about only
providing ring-back tone to the caller you can simply park the call and
use the playback app to play a tone_stream until the agent clicks the
web link, which will transfer the call from the parking extension to the
agent with the auto-answer flag.
I'm still willing to make some tests with REINVITE providing auto-answer
headers, as suggested by Mike. That would provide a more generic way to
answer calls programmatically when it's already ringing the endpoint. I
just need to find some time to read the sofia code and figure out how to
do that

Regards,

Raul

On Tue, 2009-06-16 at 02:19 +0200, Peter P GMX wrote:

[Prometheus001 at gmx.net]

It mainly works now by uuid_transfer the following way via event socket.
uuid_setvar <unique_id> sip_invite_params intercom=true
uuid_setvar <unique_id> sip_auto_answer true
uuid_transfer <unique_id> 1000 XML default
so the call is transferred from 1000 to 1000.

What happens:
1) If I disable intercom on the Snom phone, the phone rings, stops
ringing and rings again (ok)
1) If I enable intercom on the Snom phone, the phone rings, stops
ringing and hangs up (not ok)

So I do not get the Snom to pick up the call in intercom mode.

The last invite is:
INVITE sip:1000@192.168.178.50:2752;transport=tls;line=er6kxnib SIP/2.0
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
Route: <sip:1000@217.24.11.189:2752>;transport=tls;line=er6kxnib
Max-Forwards: 68
From: "Peter FS" <sip:723323@217.xx.xx.xxx>;tag=9eQ8rjQy533HF
To:
<sip:1000@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true>
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact: <sip:mod_sofia@217.xx.xx.xxx:5061;transport=tls>
Call-Info: <sip:217.xx.xx.xxx>;answer-after=0
The intercom part is there and the Call-Info line with answer-after also.

The phone answers with
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 217.xx.xx.xxx;branch=z9hG4bK3gjpa64pSKgmF
From: "Peter FS" <sip:723323@217.xx.xx.xxx>;tag=9eQ8rjQy533HF
To:
<sip:1000@192.168.178.50:2752;transport=tls;line=er6kxnib;intercom=true>;tag=71rskygkr2
Call-ID: bd05aeab-d53a-122c-72b5-001e904cc34e
CSeq: 116467629 INVITE
Contact:
<sip:1000@192.168.178.50:2752;transport=tls;line=er6kxnib>;reg-id=1
WWW-Authenticate: Digest realm="sip2.mycompany.de",
nonce="2ee26efe6ab27f88", algorithm=MD5
Content-Length: 0
and hangs up.

Anybody know how to solve this Snom intercom issue?

Best regards
Peter


Michael Jerris schrieb:

[mike at jerris.com]

uuid_setvar <unique_id> sip_invite_params intercom=true should be
unnecessary.

Mike

On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:

[Prometheus001 at gmx.net]

Thanks Michael,

I have disabled it now.

I finally got it to work, (sip_h_Call-Info=<sip:$${domain}>;answer-after=0)
but the behaviour was not as desired, as I didn't manage the phone to
pick up the call on the headset. It will only have the speaker enabled.
So I will have to go a different way with parking the call and then
forward it.

Best regards
Peter


Michael Jerris schrieb:

[anthony.minessale at g...]

<clippy>Looks like you are trying to build a call center</clippy>

have you seen mod_fifo?
It's designed to let people on headsets sit idle and you can send calls to them at will.


On Tue, Jun 16, 2009 at 3:11 PM, Peter P GMX <Prometheus001@gmx.net (Prometheus001@gmx.net)> wrote: