Call Limit 50?

General Help
1

I have an extension 102 in a queue. After awhile, ext 102 cannot receive calls and has a status of UNAVAILABLE. I entered “sip show inuse” and it has:

  • User name In use Limit
  • Peer name In use Limit
    105 32 50
    104 26 50
    103 1 50
    102 50 50
    101 35 50

My sip_additional.conf has:
[102]
type=friend
secret=??? (hidden for a reason)
record_out=Always
record_in=Always
qualify=yes
port=5060
pickupgroup=
nat=no
[email protected]
host=dynamic
dtmfmode=rfc2833
disallow=all
dial=SIP/102
context=from-internal
canreinvite=no
callgroup=
callerid=device <102>
allow=ulaw
accountcode=

In the cli, I see “chan_sip.c: 2275 update_call_counter: Call to peer ‘102’ rejected due to usage limit of 50”.

2

sounds odd. For starters, you are not showing it setup in your sip_additional.conf file where it is set if you are running Aterisk 1.4. This is required in Asterisk 1.4 along with a few other settings that come setup in sip.conf in order to have BLF and other proper functionality work. Setting the limits to 50 is effectively having them be unlimited (and allowing the device itself to throttle its own limits). As far as why it thinks you are hitting the limit, ??? If you are not running the latest 1.4 release then make sure to upgrade to it as there were known issues in this department related to these settings and this functionality.

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

3

Opps. I forgot to mention that I am running Asterisk 1.2.24. Previously, I was running Asterisk 1.2.20 and then I upgraded to Asterisk 1.4.x (I forgot which one) and then I downgraded to 1.2.24. I am running FreePBX 2.3.0 though.

My sip.conf has:
[general]
#include sip_general_additiona.conf
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw

context=from-sip-external
callerid=Unknown
tos=0x68

; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
;limitonpeers=yes ; I just commented this out but it didn’t do anything

#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf

4

then I have no idea where those call limits are coming from because as you can see in sip_additional.conf, we don’t set any lmits. Make sure that you have not set them in an include file somewhere. (And even if you have, I don’t know why/how you could be hitting them).

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

5

I created a new extension 120 and after running “sip show peer 120”, it has a call limit of 0. I will try deleting and adding the extensions later to see if it works (backing up voicemail beforehand though).

6

Ok. I performed a “restart when convenient” and it cleared up all of the call limits. The only hypothesis I can think of is that at one point call-limit was inserted into the configurations or it was the upgrading to 1.4.x and then downgrading to 1.2.x for Asterisk.

7

if you had 1.4 loaded at some point, it sounds like you restarted 1.2 with the conf files generated from 1.4. And further, it sounds like Asterisk may not be great at forgetting all of its sip configurations once set without restarting. There are some parts of Asterisk that are like that - probably because the phones have already registered.

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx