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

[Brent Davidson]

Steve Totaro wrote:
> 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.
On the subject of CallerID and ringing, I'm not sure if it's like this 
everywhere in the US, but where I live in Texas, our caller ID signal is 
sent between the first and second rings.  If the phone is answered in 
the middle of the first ring then CID signal is never received.  This 
might not be an issue in the scenario being discussed, because it sounds 
more like you're asking for Asterisk to connect the ringing Zap channel 
to a sip line before issuing an "answer" in the dialplan.  Correct me if 
I'm wrong.  I'm more used to using Asterisk in a PBX context with an 
automated attendant that answers every call before ringing any of the 
extensions. The direct Zap to Sip without without a menu is more of a 
switch context correct?

-Brent
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