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Hi
I'm using Asterisk 1.6.0.22, and have a bit non-standard setup. I have a
queue with three static agents, and the ringall strategy.
While call A is answered by agent 1, two more calls, B and C enter the
queue. Agents 2 and 3 does nothing and
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You know there are 1st of april jokes, and there are evil 1st of april
jokes.
... I actually felt a bit nauseous
- - Tommy
Den 01. april 2010 07:53, skrev Olle E. Johansson:
FOR IMMEDIATE RELEASE
Puerto Escondido, Mexico, April 1st, 2010:
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Peter den Hartog skrev:
Hello,
I was wondering if the following was possible:
When somebody sends a fax to my direct number 0101234567105 (my
extension will be 105) is it possible that Asterisk, or an addon sees
this as a fax, and e-mail the
willing to stretch on price to get decent quality.
Thanks,
Tommy Botten Jensen
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0.0.6-pre17 with
the exact same error messages. But my asterisk does not have any trouble
apart from the messages themselves.
I did however run into this earlier, and I believe it was fixed for the
1.6.2.x series. At least it worked for me.
Best regards,
Tommy Botten Jensen
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Hi
It might seem that you have installed 1.6.2.2 over 1.6.2.0, but not
updated the correct modules.
If you - from the CLI - do a mode show like sql. What is your result?
- - Tommy
ahmed magdy skrev:
Hello,
Asterisk Real time database worked
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Douglas Pasqua skrev:
I need a way to identify what client my employees are calling. For example:
- For each call that an employee of my company make to a customer, must
identify the client name in the CDR table.
- Is there a way of my
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Michelle Dupuis skrev:
We're creating a SIP gateway for a client that will take one leg of a
call in via SIP, and out the other side via H.323. To minimize load on
the gateway, we would like to have the RTP stream bypass the gatewayy
altogether
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Olle E. Johansson skrev:
23 feb 2010 kl. 20.18 skrev Matt Riddell:
The responses from the Asterisk manager on your machine start
providing responses of no account code when calls are initiated at a
higher rate.
Where's the bug report id?
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Matt Riddell skrev:
Yeah, the problem's not the origination.
The problem is that calls originated asyn with accountcodes show up in
show channels concise without details.
Pretty simple to test with sipp and core show channels concise.
I
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I have tried to replicate this, but with no luck. If I use a SIP-client
that supports '', I still get a reject from asterisk.
I am writing a filter for lua (pbx_lua), but it's a bit hard when I
cannot reproduce and test this. I've tried with both
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I'd like to jump in here as well, with the Aastra 57i. It is easy to
configure with asterisk, provision and is not that badly priced either.
- - Tommy
William Stillwell (Lists) skrev:
Polycom 331’s are also in the same price range, and offer good
that are not implemented, or known stability
issues?
Thanks,
Tommy Botten Jensen
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=TZ4p
'udptl set debug' to retrieve debug
information about the T.38 data stream.
Hope this helps.
- - Tommy Botten Jensen
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As Tzafrir mentions, the AMI can be used here. It also has a
web-interface, so you can do it via a series of URLs. Look up AJAM which
might be what you are looking for.
- - Tommy Botten Jensen
tom skrev:
hi, im trying to figure out whats
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