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Tue 02 of Dec, 2008 [14:07 UTC]

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Asterisk sip incominglimit

Created by: oej,Last modification on Thu 01 of Nov, 2007 [11:04 UTC] by neutrino38
In Asterisk 1.4

incominglimit and call-limit are equivalent. outgoinglimit is not supported anymore.

In Asterisk 1.2

incominglimit and outgoinglimit is replaced by call-limit.

Setting the call-limit will limit the number of simultaneous calls for a user or a peer.

(It does not require a username= setting, regardless of the comment below).




Comments

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222

333How to decrease

by benz, Tuesday 17 of April, 2007 [01:59:53 UTC]
Whenever we use the app SetGroup(groupname) in our dialplan,the ${GROUP_COUNT} will increase 1.
When we hangup,will the ${GROUP_COUNT} can decrease automatic ?
222

333take care of the syntax

by ViennaAustria, Tuesday 04 of April, 2006 [15:19:46 UTC]
in contrast to the description above, the parameter is not spelled call_limit (with an underscore) but call-limit (with a minus sign)!!!

so use
 call-limit=1
to allow only one incoming or outgoing call on an extension
222

333call-limit only works with type=peer

by whoiswes, Monday 27 of February, 2006 [22:18:13 UTC]
we had call-limit=1 working on our test box running 1.2.1, once patch 6111 was applied...that has been added to the source and is present in 1.2.4, as far as i can tell.

however, 1.2.4 is broken now, as it is continually setting the device state 1 (not in use). I don't see anything obvious in the source code, but then again, I'm not a programmer by trade.

but, after the patch in bug 6111 was applied, call-limit WAS working properly, and I would be upgrading all of our servers to 1.2.4 if it was working still.



EDIT: the fix is to change type=friend to type=peer in sip.conf. All of a sudden, call-limit works again. What else this might affect, I do not know, but hopefully someone with more expertise might.

this was on version 1.2.1, i'm trying 1.2.4 shortly (as soon as it's done compiling)



EDIT AGAIN: same on 1.2.4 - type=friend allows any number of calls through, while type=peer works properly with call-limit=1 (or 2, or whatever).
222

333

by whoiswes, Monday 27 of February, 2006 [21:28:51 UTC]
sorry for the dup
222

333

by jburbage, Thursday 23 of February, 2006 [23:40:46 UTC]
I am using 1.2.1 and neither call_limit nor limit works, but incominglimit still seems to work. Perhaps OEJ means that call_limit has been implemented in the 1.3 development tree (representing the upcoming 1.4 stable release).
222

333call_limit seems not to work in 1.2.4

by rszemeti, Sunday 19 of February, 2006 [23:31:58 UTC]
it seems the sip.conf parser is looking for "limit" rather than "call_limit" in version 1.0.2.4 at least, maybe others.


setting limit=1 in a sip peer, seems to do nothing at all anyway ... sip show peer <peername> shows the call limit always as zero, whatever combinations you seem to use ...

a better solution is to use the current variant of the SetGroup() command .. (which is also dprecated, along with outgoinglimit etc)



as of 1.0.2.4, this seems to work and uses current * functions


exten => _0.,1,Set(GROUP()=SOME_PROVIDER) ;Set Group
exten => _0.,2,GotoIf($${GROUP_COUNT()} > 1 ?103) ;Exceeded the 1 channel we are aloowed with this provider?
exten => _0.,3,dial(sip/8001/${EXTEN}) ; ok, dial out on SIP
exten => _0.,103,dial(${TRUNK}/g1/${EXTEN}/B,30) ; bah, ok then just fall back to our regular ISDN service.


222

333This requires username= entry

by , Monday 01 of December, 2003 [18:52:24 UTC]
See title: In order for incominglimit to work you also must have a username= entry in your sip.conf.