Pickup of a ringing extension under OpenSIPS

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Pickup of a ringing extension under OpenSIPS

Yehavi Bourvine
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).
 
                                Thanks! __Yehavi:
 

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Re: Pickup of a ringing extension under OpenSIPS

Steven C. Blair

 

Can you explain what you mean by directed pick-up?

 

-Steve

 

From: [hidden email] [mailto:[hidden email]] On Behalf Of Yehavi Bourvine
Sent: Tuesday, February 24, 2009 7:40 AM
To: Opensips
Subject: [OpenSIPS-Users] Pickup of a ringing extension under OpenSIPS

 

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).

 

                                Thanks! __Yehavi:

 


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Re: Pickup of a ringing extension under OpenSIPS

Iñaki Baz Castillo
In reply to this post by Yehavi Bourvine
2009/2/24 Yehavi Bourvine <[hidden email]>:
> 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
<[hidden email]>

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Re: Pickup of a ringing extension under OpenSIPS

Olle E. Johansson
In reply to this post by Yehavi Bourvine

24 feb 2009 kl. 13.39 skrev Yehavi Bourvine:

> 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).

If you really want future growth but still want PBX services, you  
don't want to replace Asterisk with OpenSIPS. You want to make them  
work together.

/O

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Re: Pickup of a ringing extension under OpenSIPS

Yehavi Bourvine
Steve C. Blair wrote:
 
> Can you explain what you mean by directed pick-up?
 
when I hear a nearby extension ringing I dial something which includes its number and then I am the one who is answering the call. Currently I do this with the BLF/Speed dials and asterisk.

and:
2009/2/24 Johansson Olle E <[hidden email]>

24 feb 2009 kl. 13.39 skrev Yehavi Bourvine:

 
If you really want future growth but still want PBX services, you don't want to replace Asterisk with OpenSIPS. You want to make them work together.
My idea is to let OpenSIPS do all the local work (internal extensions) and Asterisk do the more complicated work like voice mail and transcoding for external voice connections.
 
                           Thanks! __Yehavi:

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Re: Pickup of a ringing extension under OpenSIPS

Zahid Mehmood
In reply to this post by Yehavi Bourvine
If you are still using the Polycom phones, you can play with the  
"group-call-pickup", "directed-call-pickup" features.  Enabling these  
two will make two soft keys visible when you make the phone go off-
hook.  Essentially the phone will use subscribe/notify to find out  
the state and then uses refer or invite (i dont remember right now)  
to receive the call.

I suggest that you enable the above mentioned polycom "features", and  
do a packet capture to see what type of packets they generate.  Use a  
simple opensips config that relays any subscribe/notify to the RURI  
directly will help get a better picture.  Also note the user part of  
the RURI when using these features.

HTH.

--
Zahid


On Feb 24, 2009, at 7:39 AM, Yehavi Bourvine wrote:

> 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).
>
>                                 Thanks! __Yehavi:
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: Pickup of a ringing extension under OpenSIPS

Yehavi Bourvine
I am using Polycom phones. Will look into these definitions.
 
                               Thanks! __Yehavi:

2009/2/24 Zahid Mehmood <[hidden email]>
If you are still using the Polycom phones, you can play with the
"group-call-pickup", "directed-call-pickup" features.  Enabling these
two will make two soft keys visible when you make the phone go off-
hook.  Essentially the phone will use subscribe/notify to find out
the state and then uses refer or invite (i dont remember right now)
to receive the call.

I suggest that you enable the above mentioned polycom "features", and
do a packet capture to see what type of packets they generate.  Use a
simple opensips config that relays any subscribe/notify to the RURI
directly will help get a better picture.  Also note the user part of
the RURI when using these features.

HTH.

--
Zahid


On Feb 24, 2009, at 7:39 AM, Yehavi Bourvine wrote:

> 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).
>
>                                 Thanks! __Yehavi:
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: Pickup of a ringing extension under OpenSIPS

Zahid Mehmood
In reply to this post by Iñaki Baz Castillo
Forgot to include the list before....




On Feb 26, 2009, at 2:42 PM, Yehavi Bourvine wrote:
2009/2/26 Zahid Mehmood <[hidden email]>
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 <a href="sip:12345@">sip:12345@mysiphost.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.12.147;branch=z9hG4bKfdaa927d9F490C1E.
From: "Zahid" <<a href="sip:10512@">sip:10512@mysiphost.com>;tag=32392D2-480E0ECB.
To: <<a href="sip:12345@">sip:12345@mysiphost.com>.
CSeq: 1 SUBSCRIBE.
Call-ID: [hidden email].
Contact: <<a href="sip:10512@">sip:10512@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 <a href="sip:groupcallpickup@">sip:groupcallpickup@mysiphost.com SIP/2.0.
Record-Route: <<a href="sip:192.168.1.50;lr=on;ftag=D0E153EE-4A592307">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" <<a href="sip:10512@">sip:10512@mysiphost.com>;tag=D0E153EE-4A592307.
To: <<a href="sip:groupcallpickup@">sip:groupcallpickup@mysiphost.com>.
CSeq: 1 SUBSCRIBE.
Call-ID: [hidden email].
Contact: <<a href="sip:10512@">sip:10512@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 <[hidden email]>:
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
<[hidden email]>

_______________________________________________
Users mailing list
[hidden email]
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