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Re: SB Error + VOIP routing
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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
>
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<a class="moz-txt-link-freetext" href="http://mi4.biz/files/testing/20070211switchboard.zip";>http://mi4.biz/files/testing/20070211switchboard.zip</a><br>
<br>
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!<br>
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.<br>
<br>
James<br>
<br>
James Traynor wrote:
<blockquote cite="mid:45CDC33A.4000402@xxxxxxx"
type="cite">
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<title></title>
Gregg Liming wrote:
<blockquote cite="mid:45CBF0B6.9040509@xxxxxxx"
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 contact
per number we talked about. It still needs some work but it should help
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"
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?
</pre>
</blockquote>
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.<br>
You can also customise the network drop down list on the phone number
screen.<br>
<br>
<blockquote cite="mid:45CBF0B6.9040509@xxxxxxx"
type="cite">
<pre wrap=""> 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. </pre>
</blockquote>
I think it was just the fact of switching from Vpstn which it didn't
appear from to pstn where it does.<br>
<br>
<blockquote cite="mid:45CBF0B6.9040509@xxxxxxx"
type="cite">
<pre wrap="">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.
</pre>
</blockquote>
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.<br>
<br>
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.<br>
<br>
Should be a update out ina few hours<br>
<br>
James<br>
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</blockquote>
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