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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 when
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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
I'm willing to stretch on price to get decent quality.
Thanks,
Tommy Botten Jensen
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fallthrough" line
>>> the following came up:
>>> pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1'
>>>
>>>
>>> and also here's the backtracce (i'm using Debian lenny)
>>>
>>> *** glibc dete
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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 e
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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 work
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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.
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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
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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
> altoget
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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
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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
dancy,maxdatagram=400
You can also use the CLI command 'udptl set debug' to retrieve debug
information about the T.38 data stream.
Hope this helps.
- - Tommy Botten Jensen
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es that are not implemented, or known stability
issues?
Thanks,
Tommy Botten Jensen
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