Posted: Sat May 30, 2009 5:41 pm Post subject: [asterisk-users] Simplex voice on TDM410P
Hello,
I am working on a trixbox based system with a TDM410P connected to 3
phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
with some polycom and Aastra SIP phones. In general everything works.
the problem I am trying to solve is that if both parties to a call speak
at the same time one of the voices gets cut out such that the talker A
cannot hear what talker B is saying. When talker A stops talking, he/she
can then hear what talker B says. This issue occurs across all the
different phones we have set up. I have played with the OSLEC settings
in the thoughts that the echo cancellation was being a bit ambitious, to
no avail. Any recommendations on how to best troubleshoot / correct this
issue?
Thanks and Regards,
Nate
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Posted: Sat May 30, 2009 6:48 pm Post subject: [asterisk-users] Simplex voice on TDM410P
I have posted a similar problem earlier on this mailing list with my Asterisk-system + TDM410 + Grandstream telephones.
But there has not yet been a response to this.
My client is also experiencing a 'simplex' conversation. There seems that audio can only flow 1 one way at the same time.
What I have tried is change the codec on the internal SIP-network from alaw to gsm (so more compression, less bandwidth needed) but problem not yet resolved.
Also I don't know where to begin to look for the problem...
So, I'm curious for the solution.
Greetingz,
Jonas.
On Sat, 2009-05-30 at 14:35 -0400, Nathanial A. Byrnes wrote:
Quote:
Quote:
Hello,
I am working on a trixbox based system with a TDM410P connected to 3
phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
with some polycom and Aastra SIP phones. In general everything works.
the problem I am trying to solve is that if both parties to a call speak
at the same time one of the voices gets cut out such that the talker A
cannot hear what talker B is saying. When talker A stops talking, he/she
can then hear what talker B says. This issue occurs across all the
different phones we have set up. I have played with the OSLEC settings
in the thoughts that the echo cancellation was being a bit ambitious, to
no avail. Any recommendations on how to best troubleshoot / correct this
issue?
Posted: Sat May 30, 2009 8:29 pm Post subject: [asterisk-users] Simplex voice on TDM410P
jonas kellens wrote:
Quote:
I have posted a similar problem earlier on this mailing list with my
Asterisk-system + TDM410 + Grandstream telephones.
But there has not yet been a response to this.
My client is also experiencing a 'simplex' conversation. There seems
that audio can only flow 1 one way at the same time.
What I have tried is change the codec on the internal SIP-network from
alaw to gsm (so more compression, less bandwidth needed) but problem
not yet resolved.
Also I don't know where to begin to look for the problem...
So, I'm curious for the solution.
I have seen this happen with the aggressive echo cancel algorithms. You
might want to look into that.
On Sat, 2009-05-30 at 14:35 -0400, Nathanial A. Byrnes wrote:
>Hello,
> I am working on a trixbox based system with a TDM410P connected to 3
>phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
>with some polycom and Aastra SIP phones. In general everything works.
>the problem I am trying to solve is that if both parties to a call speak
>at the same time one of the voices gets cut out such that the talker A
>cannot hear what talker B is saying. When talker A stops talking, he/she
>can then hear what talker B says. This issue occurs across all the
>different phones we have set up. I have played with the OSLEC settings
>in the thoughts that the echo cancellation was being a bit ambitious, to
>no avail. Any recommendations on how to best troubleshoot / correct this
>issue?
>
> Thanks and Regards,
> Nate
>
>
------------------------------------------------------------------------
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Posted: Sun May 31, 2009 7:56 am Post subject: [asterisk-users] Simplex voice on TDM410P
On my TDM410P pci-card I have an hardware echo cancellation module (Digium VPMADT032 EC Modul).
I have set 'echocancel=yes' in my chan_dahdi.conf to activate this hardware module.
Do I now have 2 echo cancellers that are activated ? A software echo canceller and a hardware echo canceller ??
Form the documentation on DAHDI :
; Note that if any of your DAHDI cards have hardware echo cancellers,
Quote:
; then this setting only turns them on and off;
;
Quote:
echocancel=yes
;
That's al I have for echo cancellation... I thought the hardware module of Digium was of great quality ?!
Quote:
Quote:
I have seen this happen with the aggressive echo cancel algorithms. You
might want to look into that.
Posted: Sun May 31, 2009 3:21 pm Post subject: [asterisk-users] Simplex voice on TDM410P
jonas kellens wrote:
Quote:
On my TDM410P pci-card I have an hardware echo cancellation module
(Digium VPMADT032 EC Modul).
I have set 'echocancel=yes' in my chan_dahdi.conf to activate this
hardware module.
Do I now have 2 echo cancellers that are activated ? A software echo
canceller and a hardware echo canceller ??
Form the documentation on DAHDI :
; Note that if any of your DAHDI cards have hardware echo cancellers,
; then this setting only turns them on and off;
;
echocancel=yes
I don't think that would happen. Nevertheless try to disable the echo
cancel and see if the simplex voice is corrected. If it is, you might
want to open a ticket with Digium since you have a problem with the
Hardware Echo Cancel module.
Posted: Sun May 31, 2009 3:37 pm Post subject: [asterisk-users] Simplex voice on TDM410P
On Sat, May 30, 2009 at 02:35:43PM -0400, Nathanial A. Byrnes wrote:
Quote:
Hello,
I am working on a trixbox based system with a TDM410P connected to 3
phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
with some polycom and Aastra SIP phones. In general everything works.
the problem I am trying to solve is that if both parties to a call speak
at the same time one of the voices gets cut out such that the talker A
cannot hear what talker B is saying. When talker A stops talking, he/she
can then hear what talker B says. This issue occurs across all the
different phones we have set up. I have played with the OSLEC settings
in the thoughts that the echo cancellation was being a bit ambitious, to
no avail. Any recommendations on how to best troubleshoot / correct this
issue?
Is there a problem with SIP<->SIP call? I suppose there isn't and that
you've already tested that.
Here 'N' is the DAHDI channel number to dial through and NUMBER is the
number to dial.
Another thing you can do is to use dahdi_monitor to either look at the
audio levels or record the audio. You can clearly see there when there's
no audio in a certain direction. This is the audio Asterisk sends to
Zaptel and recieves from it. Note, however, that the digits that DAHDI
dials ar esents as a DAHDI_DIAL ioctl rather than an explicit digit
sound.
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