[asterisk-users] Asterisk 1.8 Transfer CallerID

[Jonas Kellens]

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 at telenet.be>
>>> To: "Asterisk Users Mailing List - Non-Commercial 
>>> Discussion"<asterisk-users at lists.digium.com>
>>> Sent: Tuesday, May 8, 2012 7:13:30 AM
>>> Subject: [asterisk-users] 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.