[VoIP] T1 related Problem

[Shane Young]

The "t" that I suggested allows the called party to initiate a  
transfer by hitting the # key on their phone, however you have it in  
the wrong spot :)

The dial string just looks plain wrong to me, not even sure how it works :)

Any chance you could put a butt set on the line and see (hear) what's  
happening different.


Quoting Lee Spenadel <lee at spenadel.com>:

> What does that "t" at the end do?  I just cut over from a Linksys ATA to my
> Adtran 850.  I couldn't get Asterisk to pulse out properly to the Step
> switch until I put that "t" in my statement:
>
> Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})
>
> After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})
>
> It now works!  I've spent the last 2 hours trying to figure out these very
> strange happenings:
>
> When there's no battery connected to my FXO channel (any one for that
> matter) I can see the pulsing on the FXO led.  The pulsing is correct for
> the extension that I dialed.  However, when I connect battery to it, such as
> an extension from my Stepper, I get consistent but strange dialing.  No
> matter what extension I try to dial in Asterisk, the Stepper sees digit 2
> and whatever the last digit of the called extension was.  For example, I
> dial 349-8630, the Stepper sees 20.  That's it.  If I dial 349-8692, Step
> sees 22.  It would always see the first digit as 2 followed by the last
> digit of the called number.  I was pulling my hair out of my head until I
> came upstairs and saw this email from Shane.  I was following this thread
> between Shane and Kirt as I'm also getting the bridge messages.
>
>   -- Attempting native bridge of Zap/1-1 and Zap/21-1
>
> But it doesn't seem to be a problem.
>
> Thanks for inadvertently helping me solve my problem guys.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Shane Young
> Sent: Sunday, February 04, 2007 10:44 PM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> Well, as I said, the bridge doesn't happen until after it's done dialing and
> has received an answer.
>
> I have to go back and look to see exactly what the problem was, but, in the
> meantime, we should be able to disable the native bridging by requesting
> Asterisk stay in the audio path all the time.
>
> Try adding a "t" to the end of the dial command like this:
> Dial(zap/xx||t)
>
>
>
>
>
>
> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>
>> Shane,
>>
>> I have to think the native bridge must be involved. This problem only
>> happens when the call originates and terminates on the T1 bank. Calls
>> coming in from either my SIP phone or CNET go out just fine to my switch.
>>
>> Kirt
>>
>> Shane Young wrote:
>>
>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>
>>>
>>>
>>>> Let me clarify oen thing.
>>>>
>>>> When I was using the TDM 400 card I never saw the message about the
>>>> native bridge,a nd the same thing worked. The problem started when I
>>>> went to T1 and stoppe dusing the tDM 400 (although it is still in
>>>> the machine).
>>>>
>>>> It almost appears to be a timing problem - perhaps asterisk is
>>>> trying the native bridging before it outpulses to the switch.
>>>>
>>>>
>>>
>>> I've never seen it either for a zap channel, however I have features
>>> enabled which would always prevent it.
>>>
>>> Again, it waits until it get's the answer back from the channel
>>> before it attempts this (as you show in your output) so I don't think
>>> the native bridge is the problem.
>>>
>>> You could try putting in a wait in your dial string before dialing to
>>> see if that helps.
>>>
>>> --Shane
>>>
>>>
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>>>
>>>
>>>
>>>
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>>
>
>
>
>
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