Steve Totaro wrote:
On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <[EMAIL PROTECTED]> wrote:
On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> 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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