asterisk users - Dec 2014 - chan_sip and 2 devices under same extension - transferring call endpoint(s)

Hi, 
(please excuse me for lack of proper jargon usage and the vagueness of
description...)

i use Asterisk 11.12.1, (well... as included in FreePBX),

I have several extensions that can register 2 separate devices (chan_sip)
( FreePBX calls this Devices & Users mode : Users are extension/internal
number,
devices are the 'SIP Accounts' for the internal 'endpoints' )

(this I'm told apparently will not be needed if I switched to chan_pjsip,
since it allows multiple devices to register on the same user/secret,
so the u/d mode would not make sense any more; however this creates another
interesting problem, pls read on)

Some endpoints are grouped in pairs so that calling an extension, rings on both
devices.

(One 'device' is a real handset, usually dumb: SPA112 or SPA301,
 the other is a softphone (CSipSimple or WebRTC or both) used to bring the
incoming CID to
 users' eye level and to perform some client-side CRM integration )

On Incoming call, as expected, the softphone shows me the CID [as intended]
and I can pick up the handset, then the softphone will stop ringing; 
This far, it works as intended and no problems here.

I *think* by the FreePBX convention (?) one can not call the 'device'
number/reg directly,
only the 'user' extension [i actually tried dialing to one of the
'device' SIP reg numbers,
'cannot be completed as dialed' was the answer, and same in the -vvvvr
output;
the -vvvvr output actually suggests one side RTP is passed, but the other is
not,
if I read this correctly (on 'normal' calls, both sides RTP is shown
'passed' in the log).

The softphones are mostly on machines without proper sound hardware (no mics, no
speakers/headsets);
This is partly because the workforce is quite conservative in what they want to
use :)
meaning handsets are important; 

As the handsets have no LCD's to show the dialled number, 
I want to give the workforce the ability to dial OUT using the softphone, 
(as in, copy/paste the number from the CRM software into softphone then
*immediately* transfer the originated call 'endpoint' to the handset of
the same 'user' extension, somehow,
the question is, HOW ?

An answer from the FreePBX forum suggested SLA / Shared Line Appearance - but
as I read description of that, it's not really: there is no master/slave in
the pair,
both devices are *supposed* to be of 'equal rights' as they are
'manned' by the
same person. IOW my use case is *simpler* than SLA...

The interesting question also is how would one do this with chan_pjsip,
if a user can have multiple devices registered on the same 'SIP
Account',
how could the user 'transfer the call endpoint' between his devices 
(whether the call is incoming or outgoing) ?

Hope the above makes (some) sense,

Kind Regards

I have the very same situation in one of my networks. 
To solve this you can dial out from the softphone and to move call to the
phone you can simply transfer call to the same user (just if you were
transferring call to yourself and the other device will ring.
While, as you notice, you cannot dial a device, you can surely call your
user to tranfer from a device to another.

Please note that call waiting has to be enable on user settings otherwise it
won't work.

This should work both on device/user with chan_sip and pjsip with multiple
devices on same user.

Davide


-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Per conto di Lukasz Sokol
Inviato: luned? 29 dicembre 2014 13:26
A: asterisk-users at lists.digium.com
Oggetto: [asterisk-users] chan_sip and 2 devices under same extension -
transferring call endpoint(s)

Hi,
(please excuse me for lack of proper jargon usage and the vagueness of
description...)

i use Asterisk 11.12.1, (well... as included in FreePBX),

I have several extensions that can register 2 separate devices (chan_sip) (
FreePBX calls this Devices & Users mode : Users are extension/internal
number, devices are the 'SIP Accounts' for the internal
'endpoints' )

(this I'm told apparently will not be needed if I switched to chan_pjsip,
since it allows multiple devices to register on the same user/secret, so the
u/d mode would not make sense any more; however this creates another
interesting problem, pls read on)

Some endpoints are grouped in pairs so that calling an extension, rings on
both devices.

(One 'device' is a real handset, usually dumb: SPA112 or SPA301,  the
other
is a softphone (CSipSimple or WebRTC or both) used to bring the incoming CID
to  users' eye level and to perform some client-side CRM integration )

On Incoming call, as expected, the softphone shows me the CID [as intended]
and I can pick up the handset, then the softphone will stop ringing; This
far, it works as intended and no problems here.

I *think* by the FreePBX convention (?) one can not call the 'device'
number/reg directly, only the 'user' extension [i actually tried dialing
to
one of the 'device' SIP reg numbers, 'cannot be completed as
dialed' was the
answer, and same in the -vvvvr output; the -vvvvr output actually suggests
one side RTP is passed, but the other is not, if I read this correctly (on
'normal' calls, both sides RTP is shown 'passed' in the log).

The softphones are mostly on machines without proper sound hardware (no
mics, no speakers/headsets); This is partly because the workforce is quite
conservative in what they want to use :) meaning handsets are important; 

As the handsets have no LCD's to show the dialled number, I want to give the
workforce the ability to dial OUT using the softphone, (as in, copy/paste
the number from the CRM software into softphone then
*immediately* transfer the originated call 'endpoint' to the handset of
the
same 'user' extension, somehow, the question is, HOW ?

An answer from the FreePBX forum suggested SLA / Shared Line Appearance -
but as I read description of that, it's not really: there is no master/slave
in the pair, both devices are *supposed* to be of 'equal rights' as they
are
'manned' by the same person. IOW my use case is *simpler* than SLA...

The interesting question also is how would one do this with chan_pjsip, if a
user can have multiple devices registered on the same 'SIP Account', how
could the user 'transfer the call endpoint' between his devices (whether
the
call is incoming or outgoing) ?

Hope the above makes (some) sense,

Kind Regards



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Hi Davide,

thanks for your answer,

On 29/12/14 16:12, Davide Anzaldi [ Net&Com ] wrote:
> I have the very same situation in one of my networks. 
> To solve this you can dial out from the softphone and to move call to the
> phone you can simply transfer call to the same user (just if you were
> transferring call to yourself and the other device will ring.
> While, as you notice, you cannot dial a device, you can surely call your
> user to tranfer from a device to another.
> 
The call will need to be *established* though, i.e. the destination needs to
answer ?
But I don't know *when* can i transfer the call... because I've no audio
on softphone
(it's on a desktop workstation PC with no speakers...)

If there was some automatic code, meaning to Asterisk 'wait until call is
established then
process the rest of dial string' ... i could add e.g. blind transfer code
after this.
> Please note that call waiting has to be enable on user settings otherwise
it
> won't work.
Thanks, noted.
> 
> This should work both on device/user with chan_sip and pjsip with multiple
> devices on same user.
> 
OK.
> Davide
Lukasz
> 
[tl; description]

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol <el.es.cr at gmail.com>
wrote:
> As the handsets have no LCD's to show the dialled number,
> I want to give the workforce the ability to dial OUT using the softphone,
> (as in, copy/paste the number from the CRM software into softphone then
> *immediately* transfer the originated call 'endpoint' to the
handset of
> the same 'user' extension, somehow,
> the question is, HOW ?
>
>
We use FreePBX and a custom CRM. What we do is use the Asterisk Manager
interface to create a call using the originate command. Asterisk dials the
users handset, once they answer Asterisk then dials the outbound number. No
need for any transferring. You could also look at Asterisk call files to
originate the call.

Ryan
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On 30/12/14 02:52, Ryan Wagoner wrote:
> 
[...]
> We use FreePBX and a custom CRM. What we do is use the Asterisk
> Manager interface to create a call using the originate command.
> Asterisk dials the users handset, once they answer Asterisk then
> dials the outbound number. No need for any transferring. You could
> also look at Asterisk call files to originate the call.
> 
Thanks ! This looks interesting too - I will check this out.
> Ryan
> 
> 
> 
Lukasz

Sounds like a job for TAPI.

Google TAPI for Asterisk or Asterisk TSP

I've been playing with SIPTAPI and it works pretty well.  It's very
simple to install and set up.

Hope this Helps

PC...

 

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Monday, December 29, 2014 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] chan_sip and 2 devices under same extension -
transferring call endpoint(s)

 

 

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol <el.es.cr at gmail.com
<mailto:el.es.cr at gmail.com> > wrote:

As the handsets have no LCD's to show the dialled number,
I want to give the workforce the ability to dial OUT using the softphone,
(as in, copy/paste the number from the CRM software into softphone then
*immediately* transfer the originated call 'endpoint' to the handset of
the same 'user' extension, somehow,
the question is, HOW ?

 

We use FreePBX and a custom CRM. What we do is use the Asterisk Manager
interface to create a call using the originate command. Asterisk dials the users
handset, once they answer Asterisk then dials the outbound number. No need for
any transferring. You could also look at Asterisk call files to originate the
call.

Ryan

 

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