[Asterisk-Users] limiting incloming call on sip phones to 1

[Anton Krall]

This by using setgroups?

 

|-----Original Message-----
|From: asterisk-users-bounces at lists.digium.com 
|[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
|pdhales at optusnet.com.au
|Sent: Sunday, November 06, 2005 5:03 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] limiting incloming call on sip 
|phones to 1
|
|If the phone has two lines on it, you can be creative and set 
|them up differently.
|(one for incoming, no limit. one for outgoing, limited to 1)
|
|PaulH
|
|----- Original Message -----
|From: "Anton Krall" <akrall-lists at intruder.com.mx>
|To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
|<asterisk-users at lists.digium.com>
|Sent: Monday, November 07, 2005 3:35 AM
|Subject: RE: [Asterisk-Users] limiting incloming call on sip 
|phones to 1
|
|
|> Hi Kebin.,
|>
|> Thx for your comments, their exactly what I read. Problem 
|comes when you
|> want to be able to make any number of incoming calls (calls 
|from the phone
|> out) but limit the number of outgoing calls (calls from 
|asterisk to the
|> phone).
|>
|>  :(
|>
|> |-----Original Message-----
|> |From: asterisk-users-bounces at lists.digium.com
|> |[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
|> |Kevin Hanson
|> |Sent: Sunday, November 06, 2005 9:09 AM
|> |To: Asterisk Users Mailing List - Non-Commercial Discussion
|> |Subject: Re: [Asterisk-Users] limiting incloming call on sip
|> |phones to 1
|> |
|> |Anton Krall wrote:
|> |
|> |>Hey Guys!
|> |>
|> |>I know sip hpones can be configured to disable call waiting
|> |but this is
|> |>for all call appearances. I was wondering if there is a 
|way to limit
|> |>outgoing calls (asterisk -> phone) to a sip phone 
|(techonology) to 1?
|> |>
|> |>Is there any other way of doing this without groups or 
|such? Any kind
|> |>of settings on sip.conf for this?
|> |>
|> |>
|> |>_______________________________________________
|> |>
|> |>
|> |You can set incominglimit=1 (1.0.x) or call-limit=1 (1.2b /
|> |CVS head) in sip.conf for that extension.
|> |
|> |These limits are named from asterisk's perspective.
|> |incominglimit is calls coming in to asterisk, so it would
|> |limit calls from the sip phone to asterisk, but not from
|> |asterisk to the phone.  outgoinglimit (1.0.x) doesn't work
|> |from what I've read.
|> |
|> |call-limit is both directions.  It may be what you need.
|> |However, you won't be able to do an attended transfer.  Blind
|> |transfer might work, but I haven't tried it.
|> |
|> |quote from previous thread from Olle Johansson:
|> |
|> |"incominglimit is replaced by call-limit. Please read 
|sip.conf.sample.
|> |
|> |Outgoinglimit has not worked for ages, so we removed it. One
|> |limit works for both incoming and outgoing calls now."
|> |
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