Re: Re: SB Error + VOIP routing

[James Traynor <james@xxxxxxx>]

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Hi,

Sounds like one of the dependencies is out of date.  Probably the
intranet ocx. Try upgrading it to the 1.3, the latest, and switchboard
should then run.

http://www.mi4.biz/modules.php?name=News&file=article&sid=141

and to make sure it's registered correctly type:
run these two commands from the Run.. box
regsvr32  /u c:\windows\system32\xapintranet.ocx
regsvr32  c:\windows\system32\xapintranet.ocx

James

darrenp_lock wrote:
> James,
>
> I have version 0.1.0.194 of xAPSwitchBoard installed, which is
> running fine. However, if I extract the exe in the zip below over it
> then run the program, nothing happens. The busy cursor appears for a
> fraction of a second and then nothing. Doesn't even survie long
> enough to appear in task manager.
>
> Any thoughts?
>
> Rgds, Darren.
>
> --- In xap_automation@xxxxxxx, James Traynor <james@...>
> wrote:
>
>> http://mi4.biz/files/testing/20070211switchboard.zip
>>
>> You can now set the network list and the list of networks that
have
>>
> the
>
>> disconnect option. I have added a 'Integration' tab which details
>>
> how to
>
>> link switchboard to various things including the asterisk callerID
>>
> http
>
>> serivce., still got to add more to it though!
>> Now if a call comes in and the number is matched against the
>>
> database
>
>> and if that contact's number has a network set then that network
is
>>
> used
>
>> to return a call not the one from the incoming message.
>>
>> James
>>
>> James Traynor wrote:
>>
>>> Gregg Liming wrote:
>>>
>>>> Quoting James Traynor (2/8/07 7:37 PM):
>>>>
>>>>
>>>>
>>>>> Gregg,
>>>>> This Switchboard also contains outgoing network
selection per
>>>>>
> contact
>
>>>>> per number we talked about. It still needs some work
but it
>>>>>
> should help
>
>>>>> with asterisk routing.
>>>>>
>>>>>
>>>> This looks really good!  A couple of questions/issues
though:
>>>>
>>>> 1) Now, I'm seeing an automatic "Disconnecting in 7
seconds"
>>>>
> message as
>
>>>> soon as the dial goes through.  This appears to be new. 
Does
>>>>
> axc need
>
>>>> to send a CTI.cmd w/ a CTI.cmd Offhook message to force it
to
>>>>
> continue?
>
>>>>
>>>>
>>> This was a feature added a while back to pstn calls to allow a
>>>
> dial to
>
>>> be aborted. I'm just finishing of the update and that has a
csv
>>>
> list
>
>>> of network types that this appears with so you can change teh
>>>
> behavior
>
>>> as needed.
>>> You can also customise the network drop down list on the phone
>>>
> number
>
>>> screen.
>>>
>>>
>>>>  FWIW: axc is now using the strategy that we discussed
where the
>>>>
> VPSTN
>
>>>> line remains declared free since it can create an
indefinite
>>>>
> number of
>
>>>> trunked calls.  I'm wondering if the previous line state
change
>>>>
> kept the
>
>>>> disconnect message from appearing--especially since the
original
>>>> behavior persists for (V)PSTN calls.
>>>>
>>> I think it was just the fact of switching from Vpstn which it
>>>
> didn't
>
>>> appear from to pstn where it does.
>>>
>>>
>>>> 2) This is truly a "nit", but if initiating a
dial from the call
>>>>
> log of
>
>>>> a number that is now associated w/ a contact, it seems
that SB
>>>>
> still
>
>>>> assumes the line type to be PSTN rather than propagate the
line
>>>>
> type
>
>>>> that is now selectable in the contact record.  In
contrast,
>>>>
> dialing via
>
>>>> the contact record is perfect.
>>>>
>>>>
>>> A dial from the call log uses whatever network was reported
when
>>>
> the
>
>>> call came in, I assumed a returned call would go back along
the
>>>
> same
>
>>> network it came in on. I will look at using the network from
the
>>> contacts db if it's set.
>>>
>>> As a side note I was playing with the new trixbox ( vmware
>>>
> version)
>
>>> and noticed a callerID lookup function. It does a DB/HTTP
lookup
>>>
> to
>
>>> collect a caller's name. Anyway I've added a url in
switchboard
>>>
> where
>
>>> you can point this too so you now get callerID names directly
>>>
> back
>
>>> into * and then on tot he phones. This is just to provide name
>>>
> info
>
>>> back to asterisk, it doesn't effect any of the call logging or
>>> tracking that is based on the axc plugin.
>>>
>>> Should be a update out ina  few hours
>>>
>>> James
>>>
>>>
>
>
>
>
>
>
> Yahoo! Groups Links
>
>
>
>
>
>


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Hi,<br>
<br>
Sounds like one of the dependencies is out of date.&nbsp; Probably the
intranet ocx. Try upgrading it to the 1.3, the latest, and switchboard
should then run.<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.mi4.biz/modules.php?name=News&file=article&sid=141";>http://www.mi4.biz/modules.php?name=News&amp;file=article&amp;sid=141</a><br>
<br>
and to make sure it's registered correctly type:<br>
run these two commands from the Run.. box<br>
<font color="black" face="Times New Roman"
size="3"><span
style="font-size: 12pt;">regsvr32&nbsp; /u
c:\windows\system32\xapintranet.ocx<br>
regsvr32&nbsp; c:\windows\system32\xapintranet.ocx<br>
<br>
James<br>
<br>
</span></font>darrenp_lock wrote:
<blockquote cite="mid:etiq10+54as@xxxxxxx"
type="cite">
<pre wrap="">James,

I have version 0.1.0.194 of xAPSwitchBoard installed, which is
running fine. However, if I extract the exe in the zip below over it
then run the program, nothing happens. The busy cursor appears for a
fraction of a second and then nothing. Doesn't even survie long
enough to appear in task manager.

Any thoughts?

Rgds, Darren.

--- In <a class="moz-txt-link-abbreviated" href="mailto:xap_automation@xxxxxxx";>xap_automation@xxxxxxx</a>,
James Traynor <a class="moz-txt-link-rfc2396E" href="mailto:james@...";>&lt;james@...&gt;</a>
wrote:
</pre>
<blockquote type="cite">
<pre wrap="">
<a class="moz-txt-link-freetext" href="http://mi4.biz/files/testing/20070211switchboard.zip";>http://mi4.biz/files/testing/20070211switchboard.zip</a>

You can now set the network list and the list of networks that have
</pre>
</blockquote>
<pre wrap=""><!---->the
</pre>
<blockquote type="cite">
<pre wrap="">disconnect option. I have added a
'Integration' tab which details
</pre>
</blockquote>
<pre wrap=""><!---->how to
</pre>
<blockquote type="cite">
<pre wrap="">link switchboard to various things including
the asterisk callerID
</pre>
</blockquote>
<pre wrap=""><!---->http
</pre>
<blockquote type="cite">
<pre wrap="">serivce., still got to add more to it though!
Now if a call comes in and the number is matched against the
</pre>
</blockquote>
<pre wrap=""><!---->database
</pre>
<blockquote type="cite">
<pre wrap="">and if that contact's number has a network set
then that network is
</pre>
</blockquote>
<pre wrap=""><!---->used
</pre>
<blockquote type="cite">
<pre wrap="">to return a call not the one from the incoming
message.

James

James Traynor wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Gregg Liming wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Quoting James Traynor (2/8/07 7:37 PM):


</pre>
<blockquote type="cite">
<pre wrap="">Gregg,
This Switchboard also contains outgoing network selection per
</pre>
</blockquote>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->contact
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">per number we talked about. It still needs
some work but it
</pre>
</blockquote>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->should help
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">with asterisk routing.

</pre>
</blockquote>
<pre wrap="">This looks really good!  A couple of
questions/issues though:

1) Now, I'm seeing an automatic "Disconnecting in 7 seconds"
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->message as
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">soon as the dial goes through.  This appears
to be new.  Does
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->axc need
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">to send a CTI.cmd w/ a CTI.cmd Offhook message
to force it to
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->continue?
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">
</pre>
</blockquote>
<pre wrap="">This was a feature added a while back to pstn
calls to allow a
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->dial to
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">be aborted. I'm just finishing of the update
and that has a csv
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->list
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">of network types that this appears with so you
can change teh
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->behavior
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">as needed.
You can also customise the network drop down list on the phone
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->number
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">screen.

</pre>
<blockquote type="cite">
<pre wrap=""> FWIW: axc is now using the strategy that we
discussed where the
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->VPSTN
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">line remains declared free since it can create
an indefinite
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->number of
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">trunked calls.  I'm wondering if the previous
line state change
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->kept the
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">disconnect message from appearing--especially
since the original
behavior persists for (V)PSTN calls.
</pre>
</blockquote>
<pre wrap="">I think it was just the fact of switching from
Vpstn which it
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->didn't
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">appear from to pstn where it does.

</pre>
<blockquote type="cite">
<pre wrap="">2) This is truly a "nit", but if
initiating a dial from the call
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->log of
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">a number that is now associated w/ a contact,
it seems that SB
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->still
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">assumes the line type to be PSTN rather than
propagate the line
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->type
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">that is now selectable in the contact record. 
In contrast,
</pre>
</blockquote>
</blockquote>
</blockquote>
<pre wrap=""><!---->dialing via
</pre>
<blockquote type="cite">
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">the contact record is perfect.

</pre>
</blockquote>
<pre wrap="">A dial from the call log uses whatever network
was reported when
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->the
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">call came in, I assumed a returned call would
go back along the
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->same
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">network it came in on. I will look at using
the network from the
contacts db if it's set.

As a side note I was playing with the new trixbox ( vmware
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->version)
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">and noticed a callerID lookup function. It
does a DB/HTTP lookup
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->to
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">collect a caller's name. Anyway I've added a
url in switchboard
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->where
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">you can point this too so you now get callerID
names directly
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->back
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">into * and then on tot he phones. This is just
to provide name
</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->info
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">back to asterisk, it doesn't effect any of the
call logging or
tracking that is based on the axc plugin.

Should be a update out ina  few hours

James

</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->