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

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stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 9:17 am Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:

Quote:

Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS
interface. As it is now, when the zap line gets a call, Asterisk
answers it and waits for the analog CID to be presented, then rings the
SIP phones with the call and the CID. There's a significant latency
involved in doing this.

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.

That way if a SIP phone user wants to wait for the CID, they can, but if
they just want to answer the phone without waiting for the CID, they can
do that too.

One might suggest that everyone wants to see the CID anyway, so why
bother? Because in some situations, the phone is not at an arms reach
and the person only starts making their way towards it when they start
to hear the ringing, so if the ringing starts before the CID is
available it is likely that by the time they have gotten to the phone,
the CID is available and yet the latency between the availability of the
call on the zap line and it being picked up at a ringing phone has been
reduced a ring or two.

b.

Paul B just posted the same issue and suggested the same thing you did
in this thread "decrease the time it takes for asterisk (fxsks) to
answer"

This was my reply to his issue and a feature request I guess.

"That brings up a question though, on a regular landline with caller ID
the phone rings right away, it just doesn't display caller ID info
until a couple of rings. Why not have that option in Asterisk?"

Thanks,
Steve T

gordon+asterisk at dro...
Guest

Posted: Wed Jun 11, 2008 9:57 am Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, 11 Jun 2008, Steve Totaro wrote:

Quote:

"That brings up a question though, on a regular landline with caller ID
the phone rings right away, it just doesn't display caller ID info
until a couple of rings. Why not have that option in Asterisk?"

Intersting idea... However, I live in a country where on a regular
landline with caller ID, the caller ID is displayed before the phone
rings, so make sure it's an option and not hard-wired...

Gordon

stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 10:18 am Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson
<gordon+asterisk at drogon.net> wrote:

Quote:

On Wed, 11 Jun 2008, Steve Totaro wrote:

Quote:

Gordon,

Yes, an option.

So in your country, the caller hears ringing for a few seconds before
the destination phone rings since caller ID is sent via analog
signaling?

Thanks,
Steve T

rj2807 at gmail.com
Guest

Posted: Wed Jun 11, 2008 10:53 am Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:

Quote:

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

jnovack at stromberg-c...
Guest

Posted: Wed Jun 11, 2008 10:55 am Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

Steve Totaro wrote:

Quote:

On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson
<gordon+asterisk at drogon.net> wrote:

Quote:

On Wed, 11 Jun 2008, Steve Totaro wrote:

Quote:

Gordon,

Yes, an option.

So in your country, the caller hears ringing for a few seconds before
the destination phone rings since caller ID is sent via analog
signaling?

Thanks,
Steve T

As a general note -
Ringback, what the caller hears, is not necessarily linked to the ring
sent to the called party, and has been true for many years, even in
electro-mechanical systems.

John Novack

--
Dog is my co-pilot

stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 12:40 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 11:55 AM, John Novack
<jnovack at stromberg-carlson.org> wrote:

Quote:

Steve Totaro wrote:

Quote:

On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson
<gordon+asterisk at drogon.net> wrote:

Quote:

On Wed, 11 Jun 2008, Steve Totaro wrote:

Quote:

Gordon,

Yes, an option.

So in your country, the caller hears ringing for a few seconds before
the destination phone rings since caller ID is sent via analog
signaling?

Thanks,
Steve T

That may be the case but it really has no bearing on this callerid=yes
issue. The issue is that Asterisk waits for caller ID info before
sending the call to the dialplan.

Landlines, at least mine, do not do this, if you wish to see the
caller ID info you have to watch the ringing phone for two to three
seconds.

Thanks,
Steve Totaro

gordon+asterisk at dro...
Guest

Posted: Wed Jun 11, 2008 12:44 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, 11 Jun 2008, Steve Totaro wrote:

Quote:

On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson
<gordon+asterisk at drogon.net> wrote:

Quote:

On Wed, 11 Jun 2008, Steve Totaro wrote:

Quote:

Gordon,

Yes, an option.

So in your country, the caller hears ringing for a few seconds before
the destination phone rings since caller ID is sent via analog
signaling?

Sometimes they get a brief ring, although a lot of the time it's possible
to pickup the call without the caller hearing any ring at all. (or you
hearing any ring if your phone displays the number before it rings and you
pick it up quick enough).

Asterisk can Answer() it more or less as soon as it sees the line reversal
and has seen the CID sent (< 1 second). Sooner with ignorecid set.

The CID is sent as a burst of 1200 baud FSK tones AIUI, but I'd have to
look that up to be sure...

BT's SIN 242 - Page 8 has the details - looks like the whole thing from
line reversal through signaling, V.23 data should be under 1 second, but
it can be up to 3 seconds - the data part says "<= 2.5 secs, typically
500ms". BT typically sends nothing (witheld), a 10 or 11 digit number, or
INTERNATIONAL....

Gordon

stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 12:47 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <rj2807 at gmail.com> wrote:

Quote:

On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:

Quote:

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.

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

stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 12:55 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:

Quote:

On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <rj2807 at gmail.com> wrote:

Quote:

On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:

Quote:

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.

Correction, seconds should read rings.

stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 1:39 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 2:30 PM, Brent Davidson
<brent at texascountrytitle.com> wrote:

Quote:

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

Not SIP necessarily, just progress into the dialplan, that includes
IVR or anything that can be put in a dialplan.

Your system using IVR still delays the call delivery into the
dialplan. If you remove the callerid=yes statement, do a before and
after and will see that the IVR picks up faster without having to wait
for caller ID.

Thanks,
Steve Totaro

stotaro at totarotechn...
Guest

Posted: Wed Jun 11, 2008 1:51 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:

Quote:

On Wed, 2008-06-11 at 13:30 -0500, Brent Davidson wrote:

Quote:

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.

It's like that here in Canada too.

Quote:

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.

Yeah. I had never thought it through that fully but indeed, that would
be what I'm talking about. I guess it never occurred to me that way
because I don't usually Answer() in my dialplan contexts anyway. I
don't think I've every really understood why I need to given that it all
seems to work without doing that.

b.

If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.

Thanks,
Steve T

brent at texascountryt...
Guest

Posted: Wed Jun 11, 2008 2:37 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

Steve Totaro wrote:

Quote:

On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:

If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.

Thanks,
Steve T

This is the first I've heard of this. I've never actually had the drop
after 30 seconds problem, but now that I know it can exist I'm curious.
Can you elaborate on why Answer has the potential to cause that problem?

Thanks,
Brent

rob at hillis.dyndns.org
Guest

Posted: Wed Jun 11, 2008 3:17 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

Steve Totaro wrote:

Quote:

If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.

Answer is the /cause/? Or do you mean it's the solution?

eric at fnords.org
Guest

Posted: Wed Jun 11, 2008 3:28 pm Post subject: [asterisk-users] SIP call, updated with CID as it becomes av

Answer() is seldom the solution.

Rob Hillis wrote:

Quote:

Steve Totaro wrote:

Quote:

If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.

Answer is the /cause/? Or do you mean it's the solution?

--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.

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