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[asterisk-users] BLF and Asterisk 1.6.0b2


 
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digium at sanguinarius...
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PostPosted: Sat Feb 09, 2008 12:06 pm    Post subject: [asterisk-users] BLF and Asterisk 1.6.0b2 Reply with quote

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.
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russell at digium.com
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PostPosted: Sat Feb 09, 2008 5:37 pm    Post subject: [asterisk-users] BLF and Asterisk 1.6.0b2 Reply with quote

Thomas Kenyon wrote:
Quote:
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.

What type of device are you subscribing to, is it another SIP phone? If so,
what is the associated configuration in sip.conf? Do you have call-limit set to
some value, or the combination of callcounter and busylevel? If so, what are
they set to? (You must have these options set for it to work)

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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digium at sanguinarius...
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PostPosted: Sat Feb 09, 2008 8:06 pm    Post subject: [asterisk-users] BLF and Asterisk 1.6.0b2 Reply with quote

Russell Bryant wrote:
Quote:
Thomas Kenyon wrote:
Quote:
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.

What type of device are you subscribing to, is it another SIP phone? If so,
what is the associated configuration in sip.conf? Do you have call-limit set to
some value, or the combination of callcounter and busylevel? If so, what are
they set to? (You must have these options set for it to work)

I have enough kit around to set the machine I'm testing 1.6.0b2 to use
the same configuration as the working machines.

I have got call-limits set, but it did occur to me that there's no
reason asterisk would know that there is only one extension on
SIP/<peername>.

The stranger thing is, on the machine that it's all working on, there is
a call-limit=4 set on every extension (from what I remember it prevented
a bug that got fixed ages ago and I didn't get round to lowering it again).
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