Hey John,

No they didn't had to do this because the audio path wasn't getting
converted. On the legacy PBX, the audio path was still analog, but
signalling from the phone to the PBX was digital (Key systems).

With VoIP, the ATA or PBX has to convert the analog audio signal into
digital signal and this is where most of the issues are occurring.

Thanks,


--
Stephan Monette

Business Development Director
Primus Business Services.
Tel.: 613-656-2980




On 11-10-11 4:19 PM, "John Van Ostrand" <[email protected]> wrote:

>Stephan,
>
>Did integrators have to do this with analog PBXs too? I don't recall
>those needing tweaking throughout the year. What's the difference between
>that PBX's FXO lines and ATAs?
>
>
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----- Original Message -----
>> Hey John,
>>
>> I used several FXO ATAs in the past and they all had different issues
>> like
>> DTMF, echo, static,... I tested Mediatrix, Quintum and they both had
>> issues that customers didn't like.
>>
>> A lot of our Asterisk PBX integrators seems to prefer using a PC with
>> Asterisk and Sangoma Analog FXO boards. They always use the hardware
>> base
>> echo canceller with the Sangoma cards.
>>
>> Because of the analog phone lines, you always need to tweak and adjust
>> audio levels on each line. You will find that you need to re-adjust
>> the
>> line levels 2-3 times a year to get ride of some issues depending on
>> the
>> weather!
>>
>> Sangoma has really good support on this matter.
>>
>> Some PBX integrators got tired of tweaking FXO ports and refuse to
>> sell
>> systems with FXO lines! They only sell VoIP or PRI-T1 services with
>> their
>> PBX. With the rates of PRI-T1 being sub $500.00 (we have a few
>> customers
>> that purchased PRI-T1 form Primus at sub $500/month!), many
>> integrators
>> pushes the customer to get rid of analog lines (and got rid of the
>> headaches too!).
>>
>> I would be curious to hear who has perfect results with FXO lines on
>> Asterisk with no re-tuning on all their installation (not just one
>> install)!
>>
>> Good luck.
>>
>> --
>> Stephan Monette
>>
>> Business Development Director
>> Primus Business Services.
>> Tel.: 613-656-2980
>>
>>
>>
>>
>> On 11-10-11 4:02 PM, "John Van Ostrand" <[email protected]> wrote:
>>
>> >Hi,
>> >
>> >We are struggling with a Cisco (nee Sipura) 8800. It's not capturing
>> >DMTF
>> >reliably on inbound calls to our automated attendant (asterisk) and
>> >it
>> >happens only with certain callers and never with other callers. The
>> >first
>> >one, two or three digits will be missing so a caller pressing 5102
>> >could
>> >get option 1, 0 or 2 instead of my extension. It also is more
>> >reliable
>> >when callers wait for silence between prompts before dialling.
>> >Flipping
>> >between AVT and inband doesn't change much. We gained better
>> >reliability
>> >by reducing the gain on the outgoing audio. When AVT is configured,
>> >we
>> >can see missing SIP event packets and the 8800 doesn't log the digit
>> >either so it's clear that it's just unable to recognize the tones.
>> >
>> >Is the 8800 reliable? What is a good solid ATA? What about a
>> >Mediatrix
>> >1204? Any other suggestions? We need a 4 port FXO ATA.
>> >
>> >Thanks,
>> >
>> >--
>> >John Van Ostrand
>> >CTO, co-CEO
>> >Net Direct Inc.
>> >564 Weber St. N. Unit 12, Waterloo, ON N2L 5C6
>> >Ph: 866-883-1172 x5102
>> >Fx: 519-883-8533
>> >
>> >Linux Solutions / IBM Hardware
>> >
>> >
>> >---------------------------------------------------------------------
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>> >For additional commands, e-mail: [email protected]
>> >
>>
>>
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>
>--
>John Van Ostrand
>CTO, co-CEO
>Net Direct Inc.
>564 Weber St. N. Unit 12, Waterloo, ON N2L 5C6
>Ph: 866-883-1172 x5102
>Fx: 519-883-8533
>
>Linux Solutions / IBM Hardware
>


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