Displaying 8 results from an estimated 8 matches for "busylevel".
2015 Aug 12
2
Busy level in Asterisk 11
Hi
I need to set the number of incoming calls to one, but the outgoing calls
should be unlimited. I think the busylevel parameter is for it(incoming
calls), but not works. My config is:
cat sip.conf
[general]
[template](!)
qualify=yes
cc_agent_policy=generic
cc_monitor_policy=generic
call-limit=2
busylevel=1
callcounter=yes
subscribecontext = hint
allowsubscribe=yes
[100](template)
type=friend
context=default
hos...
2008 Nov 03
0
busylevel question
I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for
testing. In addition I register a zoiper SIP soft phone.
For the Grandstream I have busylevel=1 in sip.conf.
If I place a call from the GXP280 to zoiper and then put that call on hold
from the zoiper side and then call GXP280's extension, asterisk indicates
the phone is ringing. As the GXP280 is a single line phone it does not ring
the second call. I would have expected the call to get...
2008 Nov 06
0
Asking again about busylevel
I sent this email a few days ago but did not see any responses to it:
> I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for
> testing. In addition I register a zoiper SIP soft phone.
>
> For the Grandstream I have busylevel=1 in sip.conf.
>
> If I place a call from the GXP280 to zoiper and then put that call on hold
> from the zoiper side and then call GXP280's extension, asterisk indicates the
> phone is ringing. As the GXP280 is a single line phone it does not ring the
> second call. I would have...
2009 Jul 15
4
DEVICE_STATE() and Asterisk 1.6.0.10
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than "NOT_INUSE".
I have two extensions: 6666 and 6668. I used 6668 to make a call to
yet another phone, so I know that it's busy. I then use 6666 to call
6668 and in the dialplan have a noop to see what
2011 May 02
1
sip busy detect
...my mistake could you please help me to figure out. I have added following options in my sip.conf
[7527]
type=friend
context=from-sip
host=dynamic
dtmfmode=rfc2833
callerid="Guest" <7527>
mailbox=7527 at default
nat=no
qualify=yes
cc_agent_policy=generic
cc_monitor_policy=generic
busylevel=1
limitonpeers=yes
call-limit=1
when 7527 is busy i am getting following error message on CLI. Why i am getting channel status CONGESTION ? instead BUSY ?
[May 2 16:47:31] NOTICE[31757]: chan_sip.c:5658 update_call_counter: Call to peer '7527' rejected due to usage limit of 1
-- Co...
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
...disallow: NULL
allow: NULL
insecure: NULL
trustrpid: NULL
progressinband: NULL
promiscredir: NULL
useclientcode: NULL
accountcode: NULL
setvar: NULL
callerid: NULL
amaflags: NULL
callcounter: NULL
busylevel: NULL
allowoverlap: NULL
allowsubscribe: NULL
videosupport: NULL
maxcallbitrate: NULL
rfc2833compensate: NULL
mailbox: NULL
session-timers: NULL
session-expires: NULL
session-minse: NULL
session-refresher: NULL
t38pt_usertpsource: NULL
regexten...
2008 Feb 09
1
BLF and Asterisk 1.6.0b2
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy
hints to phones?
I'm not reporting this a s a bug because (although I have it working
with Asterisk 1.4.17, the hardware involved is different.
Thanks.
2010 Jun 04
1
originating a sip call from the CLI
Hello again!
I just got a SIP account and it seems - from a config on the net -, that
I've configured it correctly. But I get no call to the outside. Registration
was OK.
I tried:
channel originate sip/1/echo at iptel.org Application ...
I see the channel active for a while, but no call gets established.
In my config I have defined the section [iptel] for the outgoing call and I