[VoIP] T1 related Problem

Mon Feb 5 07:41:12 CST 2007

I got that.

What I'm wondering is *what* is it doing to make it work :)

If you have the "t" there, can you hit # on the called phone and get  
something out of asterisk?  If not, then I want to dig deeper to see  
what the "t" does when it's in the position you put it.

Quoting Lee Spenadel <lee at spenadel.com>:

> I removed my "t" again this AM.  It broke dialing to the step switch.  Put
> it back and it worked again.
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Shane Young
> Sent: Monday, February 05, 2007 8:33 AM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> Quoting Lee Spenadel <lee at spenadel.com>:
>
>> It may be wrong, but it works just fine :P
>>
>> I don't see why this shouldn't work.
>
> Cool, maybe the t is "working" a different way :)
>
> To do what I described, the dial statement should look like this:
> _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4}||t)
>
> When the called phone answeres, can they hit # and get a prompt back from
> Asterisk? If so, then it's working as I described and is some hidden feature
> that allows you to use it where you put it rather than where the
> documentation states to put it (*CLI> show application dial)
>
>
>
>> Yes, I did put a butt set on the line and listened.  It sounded just
>> like what was being pulsed, the number 2 + the last digit dialed.
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
>> Of Shane Young
>> Sent: Monday, February 05, 2007 1:02 AM
>> To: voip at ckts.info
>> Subject: Re: [VoIP] T1 related Problem
>>
>> 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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