[OpenSIPS-Users] Pickup of a ringing extension under OpenSIPS

[Zahid Mehmood]

Forgot to include the list before....




On Feb 26, 2009, at 2:42 PM, Yehavi Bourvine wrote:
> 2009/2/26 Zahid Mehmood <zm23 at columbia.edu>
> i tested directed pickup and it worked fine in pure sip  
> environment.. the only issues i had were with the cisco media  
> gateway not working properly with  REFER etc.  Basically, instead of  
> using openser module to keep track of the dialog, the phone A (used  
> to pick) subscribes directly to the phoneB (ringing phone) for is  
> dialog state, receives the notify and then acts on it.  Trust me..  
> it would take long to get the basic working.. it just works.  There  
> will be more work if you want to implement authorization and other  
> security measures.
>
> How do you do that?



I had meant to say that it "won't take long to get the basic working".

For testing, start with the basic opensips configuration that does not  
use a presence server.

Enable feature.11.name="group-call-pickup" and feature. 
12.name="directed-call-pickup" in Polycom phone configuration.  After  
rebooting the phones, now if the phone goes off-hook, you will see a  
new soft key option "pickup".  Pressing pickup will bring you to  
another screen where you enter the ringing extension and press the  
"Directd" soft key.

Pressing that would generate a subscribe to that number.   Sample  
packet:

Directed call pickup:

U 2009/02/27 10:24:58.028551 192.168.12.147:5060 -> 192.168.1.50:5060
SUBSCRIBE sip:12345 at mysiphost.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.12.147;branch=z9hG4bKfdaa927d9F490C1E.
From: "Zahid" <sip:10512 at mysiphost.com>;tag=32392D2-480E0ECB.
To: <sip:12345 at mysiphost.com>.
CSeq: 1 SUBSCRIBE.
Call-ID: 2ba849b7-7f9ea388-e763efd1 at 192.168.12.147.
Contact: <sip:10512 at 192.168.12.147>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,  
NOTIFY, PRACK, UPDATE, REFER.
Event: dialog.
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.3.0401.
Accept: application/dialog-info+xml.
Max-Forwards: 70.
Expires: 0.
Content-Length: 0.
.

Now, if you opensips config simply relays any "subscribe" and  
"notify" (or only with event: dialog) you will be able to retrieve the  
call ringing on "12345" from the phone "10512".


Here is a sample subscribe for "group call pickup".  In this case your  
opensips config should detect the special  username (groupcallpickuy)  
in the ruri and take action to build and append branches that make  
that group.  Parallel fork to all those branches and the ringing call  
will be answered from your phone.  If more than one phone is ringing  
in the group, then the first to reply will be answered.


Group call pickup:

U 2009/02/27 10:25:11.524118 192.168.1.50:5060 -> 128.59.62.26:5060
SUBSCRIBE sip:groupcallpickup at mysiphost.com SIP/2.0.
Record-Route: <sip:192.168.1.50;lr=on;ftag=D0E153EE-4A592307>.
Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bK19e.5fcfad05.0.
Via: SIP/2.0/UDP 192.168.12.147;branch=z9hG4bK8314c4f96A1866BA.
From: "Zahid" <sip:10512 at mysiphost.com>;tag=D0E153EE-4A592307.
To: <sip:groupcallpickup at mysiphost.com>.
CSeq: 1 SUBSCRIBE.
Call-ID: c1f9ab73-3096f364-b277dccd at 192.168.12.147.
Contact: <sip:10512 at 192.168.12.147>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,  
NOTIFY, PRACK, UPDATE, REFER.
Event: dialog.
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.3.0401.
Accept: application/dialog-info+xml.
Max-Forwards: 69.
Expires: 0.
Content-Length: 0.
.

Once presence_dialoginfo and pua_dialoginfo become stable, you can  
start using that to process all the subscribe/notify messages instead  
of relying on the endpoints.

Hope this helps.

-- 
Zahid

On Feb 24, 2009, at 7:48 AM, Iñaki Baz Castillo wrote:

> 2009/2/24 Yehavi Bourvine <yehavi.bourvine at gmail.com>:
>> Hello,
>>
>>   I am in the process of duplicating my Asterisk system into  
>> OpenSIPS in
>> order to allow for a future growth. I need to do directed pickup when
>> another extension rings. How do I do that? (assuming I know who  
>> wants to
>> pickup what).
>
> OpenSIPS has a new module presence_dialoginfo and pua_dialoginfo,
> implementing RFC 4235 which allows call pick-up and so, but it doesn't
> work very well for now.
>
> If Asterisk is in the middle of the calls then you can remain using
> Asterisk PickUp (very poor anyway, but it "works").
>
> -- 
> Iñaki Baz Castillo
> <ibc at aliax.net>
>
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> Users at lists.opensips.org
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