asterisk users - Apr 2004 - Limiting incoming SIP calls & Original CallerID on transfer

Hi Erik,

I have 2 issues which I need to resolve on our production Asterisk
server:


We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an audible beep is heard over
the phone (regular call waiting). I would like to limit the number of
calls sent to each phone to 1 call only; otherwise respond as being
busy. I have looked at trying to accomplish this in the sip.conf by
using the 'incominglimit' and 'outgoinglimit' parameters,
however, the
only one that *seems* to work is the 'incominglimit'. This prevents
further calls from reaching the phones, rings busy, but does not allow
our phones to initiate a 2nd call OR transfer their existing call. The
'outgoinglimit' parameter does not seem to have any effect on limiting
whatsoever. Is there a way to limit calls passed to the phones from
Asterisk and also allow each phone to initiate 2 calls or transfer calls
(disable call waiting)??

I have also looked at the WIKI for the parameters listed above and it
*appears* that 'outgoinglimit' should do what I want, however it also
states that this function has been disabled??

"The _outgoinglimit__ is currently disabled in the source code of the
SIP channel."
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit



My second problem is that when external calls are transferred by our
receptionist to other staff members, the CallerID of course changes to
her Name instead of the original caller. Is there a way (in the
extensions logic or other) to preserve this CallerID information so that
staff members receive calls with the proper CallerID information?


Thanks,


-- 
Erik Barker

Look here:
    http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD

Jim

James H. Thompson
jht@lava.net

----- Original Message ----- 
From: "tmpm" <tmpm@softhome.net>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, April 21, 2004 11:50 AM
Subject: [Asterisk-Users] Ok, Im confused

> Im totally a newbee at *
> 
> Im confused.
> Ive got a FWD account, and it works on the winboxen. Ive got * up and can 
> do the echotest etc, so its working.
> 
> I want to get FWD working, and all the pages ive seen on setup are most 
> confusing.
> Is FWD setup like IAXTEL? Do i plug in my FWD info in the same places as 
> the IAXTEL stuff?
> Ive been trying for a week now, and Im more lost than before.
> 
> Ive got a Internet phonejack card in the penguin, phone0, and all I want to
> do at this point is make and receive calls thru FWD using that jack....Ill 
> plug the house in later...Ill learn the other stuff later. No voicemail, no
> BS, no dial thru least cost routing, or nightlines.... just make it work as
> a phone.
> 
> Im either more stupid than I think, or Im missing something major here.
> 
> Ive got to the point the CLI shows me connected to FWD fine.(I think)
> Sip show users
> 
> Username Secret Authen Def. Context a/c
> fwd.pulver.com secret md5,plaintext default no
> 
> Need some basic, stupidly simple scripts here...I dont need or want to dial
> 1-700 or *9 or any other crap, just make it work like the stupid winbox 
> phone for now...Ill read the wiki for a couple years, and then maybe I can 
> do voicemail or whatever...
> 
> frustrated...and I know its showing...sri
> 
> _______________________________________________
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>

From: "Erik Barker" <erikb@netnation.com>
> We are currently using Polycom IP600 VOIP phones for our office which
> are capable of handling 2 calls per SIP registration. What we're
finding
> is when staff are on the phone, Asterisk will pass them a second call
> which will show up on their display, and an audible beep is heard over
> the phone (regular call waiting). I would like to limit the number of
> calls sent to each phone to 1 call only; otherwise respond as being
> busy. I have looked at trying to accomplish this in the sip.conf by
> using the 'incominglimit' and 'outgoinglimit' parameters,
however, the
> only one that *seems* to work is the 'incominglimit'. This prevents
> further calls from reaching the phones, rings busy, but does not allow
> our phones to initiate a 2nd call OR transfer their existing call. The
> 'outgoinglimit' parameter does not seem to have any effect on
limiting
> whatsoever. Is there a way to limit calls passed to the phones from
> Asterisk and also allow each phone to initiate 2 calls or transfer calls
> (disable call waiting)??
>
> I have also looked at the WIKI for the parameters listed above and it
> *appears* that 'outgoinglimit' should do what I want, however it
also
> states that this function has been disabled??
>
> "The _outgoinglimit__ is currently disabled in the source code of the
> SIP channel."
>
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit

Hi to all,

Sorry for not responding a lot earlier to the issue of incominglimit as I
have been absolutely flat chat, no excuses however, hope I can shed some
light on this issue. When I first worked on this, "incominglimit" and
"outgoinglimit" was already in place, but it didn't work in
preventing call
waiting at all. Customers of ours were on GS phones, and all of you know the
problem with that on call waiting. The trouble is, with call waiting you
can't really separate incoming and outgoing cleanly. The main aim was to
prevent call waiting whenever the phone was in use (it should be changed to
"inuse" as in the CLI command "sip show inuse") whether on
an incoming or
outgoing call. Once you separate the two functions, call waiting will fail,
ie when on outgoing call, the incoming counter will not increment, therefore
you WILL receive an incoming call and vice-versa. I know that what I've done
is not pretty (ie to incorporate the two into one), but it does fix the
problem with call waiting on SIP phones that can't or won't handle it. I
am
open to ideas or suggestions and will put in the time to fix this properly
once an for all. Please reply either on or off list.

Paul