Re: Asterisk 1.8 Transfer CallerID

[Jonas Kellens <jonas.kellens@xxxxxxxxxx>]

On 05/08/2012 04:24 PM, Kevin P. Fleming wrote:
> On 05/08/2012 08:50 AM, Jonathan Rose wrote:
> > ----- Original Message -----
> > > From: "Jonas Kellens"<jonas.kellens@xxxxxxxxxx>
> > > To: "Asterisk Users Mailing List - Non-Commercial 
> > > Discussion"<asterisk-users@xxxxxxxxxxxxxxxx>
> > > Sent: Tuesday, May 8, 2012 7:13:30 AM
> > > Subject:  Asterisk 1.8 Transfer CallerID
> > >
> > >
> > > Hello,
> > >
> > > when a call comes in and is answered by colleague A, this colleague A
> > > sees the CallerID of the external calling number.
> > >
> > > When colleague A transfers the call to colleague B, attended or
> > > unattended, then colleague B sees the number of colleague A on his
> > > screen while talking to the external calling number.
> >
> > That would be because this is the expected behavior.  The call isn't
> > coming from the outside caller, it's coming from the person who
> > transferred it.
> >
> > > I expect here that colleague B would see the external calling number
> > > on the screen of his IP-phone.
> > >
> > > How can I get this behaviour ?
> > >
> > >
> > > Thanks.
> > > Jonas.
> >
> > Getting this behavior shouldn't be too hard I wouldn't think. First,
> > be aware that the Dial command has an option s(x) which is described:
> >
> > s(x): Force the outgoing callerid tag parameter to be set to the
> >      string<x>.
> >      Works with the f option.
> >
> > So if you simply transfer to a dial application with that option,
> > you can force the callerid to be whatever you want it to be. You can
> > also retrieve the callerid of the original caller and put it on your
> > transferring peer in a variable when starting the call. I'm not
> > exactly sure on the specifics of that right now, but I'm pretty sure
> > it should be possible. So then when you are making the transfer to
> > dial, you just use that variable as your argument to the s option.
>
> This is overkill, although it is certainly a way to approach it.
>
> If the OP's SIP peers for his phones are configured to send 
> Remote-Party-ID or P-Asserted-Identity to those phones
This is a setting in sip.conf ?


> This can be tested by using the CONNECTEDLINE() dialplan function to 
> send anything desired to a phone that is in a call with Asterisk.
According to the wiki it works on Polycom but not on Grandstream.

Can you tell me what exactly needs to be supported on the IP-phone ? Is 
it a certain RFC or something else ?


Jonas.


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