[asterisk-users] SIP call, updated with CID as it becomes available

[Steve Totaro]

On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
> On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <rj2807 at gmail.com> wrote:
>> On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:
>>> I'm wondering if the SIP lines can start ringing as soon as the zap line
>>> gets a call and when the zap line finally gets the CID, that is passed
>>> down to the already ringing SIP phones.
>>
>> This is actually an interesting problem. The SIP protocol didn't
>> originally support this notion, but a recent extension to SIP adds
>> this capability to the protocol. This concept is known as
>> Connected-Identity in SIP and is defined in RFC 4916. The idea is to
>> be able to update remote party's identity in either direction after
>> the call has been answered or while it is ringing. I don't think
>> people were really aware of the scenario that you've described, but it
>> is an interesting one and I think RFC 4916 covers it.
>>
>> The thing though is that even if somebody added this capability to
>> Asterisk, you'll need SIP phones that support this capability as well.
>> Right now, I don't think there is any SIP phone out there that
>> supports this.
>>
>> --
>> Raj Jain
>>
>
> If you search the archives, you will see this topic come up again and
> again, and in reality it is an issue.  If nobody answers a phone in
> say five to ten seconds (including voicemail), I hangup.
>
> Ok, then build it in now.  Make it work for DAHDI and when the phones
> start implementing the capability, Asterisk will be ready.  People
> with channel banks or similar can benefit immediately.
>
> Thanks,
> Steve Totaro
>

Correction, seconds should read rings.