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<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2>That sounds like the most practical solution....Except its
not clear to me when the number of "group unit" is actually
decreased. Does it work automagically when the call is hung up, or is
there a command to decrease it?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2>The wiki page doesnt seem to mention it ( <A
href="http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup">http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup</A> ),
its probably obvious for most people but not to me.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2>Mike</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=984183620-20112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of
</B>tracinet<BR><B>Sent:</B> November 20, 2006 1:40 PM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] Call limits and VoIP providers<BR></FONT><BR></DIV>
<DIV></DIV>I would use the Set(GROUP()=groupname) command followed by a
GotoIf($[ ${GROUP_COUNT() blah blah blah] statement that directs the
caller to a priority based on the number of channels in use. Works very
well...<BR><BR>Pedro <BR><A
href="http://www.TRACI.net">http://www.TRACI.net</A><BR>Offering SIP
Origination and Termination Services<BR><BR><BR>
<DIV><SPAN class=gmail_quote>On 11/20/06, <B
class=gmail_sendername>mail-lists</B> <<A
href="mailto:mail-lists@peachnet.com"> mail-lists@peachnet.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">this
is from freepbx though coceptually should hold true.. <BR><BR>can
you not come up with a custom dial route (ie dialing 9 or whatever)<BR>and
then limit the channels on that route to 4<BR>then have that extension send
a prefix (9) whener it's being used to<BR>access that route? <BR>> That
doesn't work thought, because calls from extension to extension<BR>>
(i.e. 201 to 205) shouldn't be taken into account. Only calls
to/from<BR>> my VoiP providers should.<BR>><BR>>
Mike<BR>><BR>>
------------------------------------------------------------------------<BR>>
*From:* <A
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A><BR>>
[mailto: <A
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A>]
*On Behalf Of *Vicky<BR>> *Sent:* November 20,
2006 12:31 PM<BR>> *To:* Asterisk Users Mailing
List - Non-Commercial Discussion <BR>> *Subject:*
Re: [asterisk-users] Call limits and VoIP
providers<BR>><BR>> You can set maximum
channels per extension in sip.conf like
using<BR>> call-limit . From
wiki<BR>> call-limit = number : Number of
simultaneous calls through this <BR>>
user/peer.<BR>><BR>> On 20/11/06, *Mike*
<<A href="mailto:list@virtutel.ca">list@virtutel.ca</A> <mailto:<A
href="mailto:list@virtutel.ca">list@virtutel.ca</A>>><BR>>
wrote:<BR>> <BR>>
Hi,<BR>><BR>> I have a
few customers hosted on my Asterisk
PBX. All<BR>>
incoming calls are coming from VoIP provider A and
all<BR>> outgoing calls
are going to termination provider B (they
<BR>> happen to be
different for
now).<BR>><BR>> On ONE
of those customers I'd like to setup a channel
limit,<BR>> meaning they
can have 4 calls max, either outgoing,
incoming,<BR>> or a
combinaison of both. After that, I want it to ring busy.
<BR>><BR>> How do I
accomplish that? I don't want to setup a global
call<BR>> limit for the
provider, because other customers are on
taking<BR>> calls and
have no limits. What is the best practice here?
<BR>><BR>>
Mike<BR>><BR>><BR>><BR>><BR>>
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