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lethol at gmail.com Guest
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Posted: Sat Apr 05, 2008 11:48 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
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|
Hi,
I've used all kinds of digium cards without troubles. My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high. The
call basically feels as in a half-duplex channel, but only to the
person behind our asterisk. I found a quick way to recreate by
placing a call using zapata channel, someplace that has an audio
stream (ie. music on hold from another pbx). When one talks into the
phone, one can notice the incoming audio getting muted until you stop
talking.
First I thought it had to do with polycom configuration although we
use the same setup for all installations (VAD, etc), but the same
happens with other sip phones and after more tests I can only recreate
this using the TDM2400p's FXO trunks. I have an older TDM2400p with
no VPMADT032 in production (without this problem), this leads me to
believe there maybe something wrong with VPMADT032 module or with my
card in particular.
Today I rebuilt everything from scratch using latest asterisk 1.2
release, rechecked with the TDM2400p manual zapata configs just to
make sure I wasn't missing something. As the manual suggests, I am
just using echocancel=yes and this should set 128 default value for
the card. In the general zapata options there we have
echocancelwhenbridged=yes. I have played with all yes/no combinations
without luck.
Interrupts and timing stuff are OK, we have good incoming and outgoing
audio quality (as long as its not at the same time).
Anyone else using this card showing the same problems?
Any zaptel/asterisk gurus wanna take a shot at this?
Thanks in advance for your feedback/comments.
Lex |
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ruben.zamora at zys.co... Guest
|
Posted: Sun Apr 06, 2008 12:48 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Hi,
I have a same problem, last week i was working with TE120 with a little
echo in some call, I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no
more echo in my call.
But know i have de same probelm with my incoming audio stream gets
clipped / dropped when you speak.
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Hi,
I've used all kinds of digium cards without troubles. My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high. The
call basically feels as in a half-duplex channel, but only to the
person behind our asterisk. I found a quick way to recreate by
placing a call using zapata channel, someplace that has an audio
stream (ie. music on hold from another pbx). When one talks into the
phone, one can notice the incoming audio getting muted until you stop
talking.
First I thought it had to do with polycom configuration although we
use the same setup for all installations (VAD, etc), but the same
happens with other sip phones and after more tests I can only recreate
this using the TDM2400p's FXO trunks. I have an older TDM2400p with
no VPMADT032 in production (without this problem), this leads me to
believe there maybe something wrong with VPMADT032 module or with my
card in particular.
Today I rebuilt everything from scratch using latest asterisk 1.2
release, rechecked with the TDM2400p manual zapata configs just to
make sure I wasn't missing something. As the manual suggests, I am
just using echocancel=yes and this should set 128 default value for
the card. In the general zapata options there we have
echocancelwhenbridged=yes. I have played with all yes/no combinations
without luck.
Interrupts and timing stuff are OK, we have good incoming and outgoing
audio quality (as long as its not at the same time).
Anyone else using this card showing the same problems?
Any zaptel/asterisk gurus wanna take a shot at this?
Thanks in advance for your feedback/comments.
Lex
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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creslin at digium.com Guest
|
Posted: Mon Apr 07, 2008 3:11 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Ruben Zamora wrote:
Quote: | Hi,
I have a same problem, last week i was working with TE120 with a little
echo in some call, I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no
more echo in my call.
But know i have de same probelm with my incoming audio stream gets
clipped / dropped when you speak.
|
Please contact Digium technical support about this. This is definitely
something that we need to work with the vendor of the echo canceller IP
about.
Matthew Fredrickson
Quote: |
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Hi,
I've used all kinds of digium cards without troubles. My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high. The
call basically feels as in a half-duplex channel, but only to the
person behind our asterisk. I found a quick way to recreate by
placing a call using zapata channel, someplace that has an audio
stream (ie. music on hold from another pbx). When one talks into the
phone, one can notice the incoming audio getting muted until you stop
talking.
First I thought it had to do with polycom configuration although we
use the same setup for all installations (VAD, etc), but the same
happens with other sip phones and after more tests I can only recreate
this using the TDM2400p's FXO trunks. I have an older TDM2400p with
no VPMADT032 in production (without this problem), this leads me to
believe there maybe something wrong with VPMADT032 module or with my
card in particular.
Today I rebuilt everything from scratch using latest asterisk 1.2
release, rechecked with the TDM2400p manual zapata configs just to
make sure I wasn't missing something. As the manual suggests, I am
just using echocancel=yes and this should set 128 default value for
the card. In the general zapata options there we have
echocancelwhenbridged=yes. I have played with all yes/no combinations
without luck.
Interrupts and timing stuff are OK, we have good incoming and outgoing
audio quality (as long as its not at the same time).
Anyone else using this card showing the same problems?
Any zaptel/asterisk gurus wanna take a shot at this?
Thanks in advance for your feedback/comments.
Lex
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc. |
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lethol at gmail.com Guest
|
Posted: Mon Apr 07, 2008 8:57 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Ruben,
Contact support at digium they have a release on a firmware that fixes
this and other issues with the VPMADT032.
Apparently it comes on newer zaptel drivers.
Good luck with your install.
Lex
On Mon, Apr 7, 2008 at 3:11 PM, Matthew Fredrickson <creslin at digium.com> wrote:
Quote: | Ruben Zamora wrote:
Quote: | Hi,
I have a same problem, last week i was working with TE120 with a little
echo in some call, I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no
more echo in my call.
But know i have de same probelm with my incoming audio stream gets
clipped / dropped when you speak.
|
Please contact Digium technical support about this. This is definitely
something that we need to work with the vendor of the echo canceller IP
about.
Matthew Fredrickson
Quote: |
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Hi,
I've used all kinds of digium cards without troubles. My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high. The
call basically feels as in a half-duplex channel, but only to the
person behind our asterisk. I found a quick way to recreate by
placing a call using zapata channel, someplace that has an audio
stream (ie. music on hold from another pbx). When one talks into the
phone, one can notice the incoming audio getting muted until you stop
talking.
First I thought it had to do with polycom configuration although we
use the same setup for all installations (VAD, etc), but the same
happens with other sip phones and after more tests I can only recreate
this using the TDM2400p's FXO trunks. I have an older TDM2400p with
no VPMADT032 in production (without this problem), this leads me to
believe there maybe something wrong with VPMADT032 module or with my
card in particular.
Today I rebuilt everything from scratch using latest asterisk 1.2
release, rechecked with the TDM2400p manual zapata configs just to
make sure I wasn't missing something. As the manual suggests, I am
just using echocancel=yes and this should set 128 default value for
the card. In the general zapata options there we have
echocancelwhenbridged=yes. I have played with all yes/no combinations
without luck.
Interrupts and timing stuff are OK, we have good incoming and outgoing
audio quality (as long as its not at the same time).
Anyone else using this card showing the same problems?
Any zaptel/asterisk gurus wanna take a shot at this?
Thanks in advance for your feedback/comments.
Lex
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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ruben.zamora at zys.co... Guest
|
Posted: Mon Apr 07, 2008 9:22 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
By the moment i have a big problem.
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Ruben,
Contact support at digium they have a release on a firmware that fixes
this and other issues with the VPMADT032.
Apparently it comes on newer zaptel drivers.
Good luck with your install.
Lex
On Mon, Apr 7, 2008 at 3:11 PM, Matthew Fredrickson <creslin at digium.com> wrote:
Quote: | Ruben Zamora wrote:
Quote: | Hi,
I have a same problem, last week i was working with TE120 with a little
echo in some call, I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no
more echo in my call.
But know i have de same probelm with my incoming audio stream gets
clipped / dropped when you speak.
|
Please contact Digium technical support about this. This is definitely
something that we need to work with the vendor of the echo canceller IP
about.
Matthew Fredrickson
Quote: |
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Hi,
I've used all kinds of digium cards without troubles. My last
installation is using a TDM2400p with VPMADT032 echo cancel module and
after a week of use we noticed that any incoming audio stream gets
clipped / dropped when you speak or when ambient noise is high. The
call basically feels as in a half-duplex channel, but only to the
person behind our asterisk. I found a quick way to recreate by
placing a call using zapata channel, someplace that has an audio
stream (ie. music on hold from another pbx). When one talks into the
phone, one can notice the incoming audio getting muted until you stop
talking.
First I thought it had to do with polycom configuration although we
use the same setup for all installations (VAD, etc), but the same
happens with other sip phones and after more tests I can only recreate
this using the TDM2400p's FXO trunks. I have an older TDM2400p with
no VPMADT032 in production (without this problem), this leads me to
believe there maybe something wrong with VPMADT032 module or with my
card in particular.
Today I rebuilt everything from scratch using latest asterisk 1.2
release, rechecked with the TDM2400p manual zapata configs just to
make sure I wasn't missing something. As the manual suggests, I am
just using echocancel=yes and this should set 128 default value for
the card. In the general zapata options there we have
echocancelwhenbridged=yes. I have played with all yes/no combinations
without luck.
Interrupts and timing stuff are OK, we have good incoming and outgoing
audio quality (as long as its not at the same time).
Anyone else using this card showing the same problems?
Any zaptel/asterisk gurus wanna take a shot at this?
Thanks in advance for your feedback/comments.
Lex
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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tzafrir.cohen at xorco... Guest
|
Posted: Tue Apr 08, 2008 1:52 am Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
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|
On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
|
Generally you can always use a newer zaptel.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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lethol at gmail.com Guest
|
Posted: Tue Apr 08, 2008 2:21 am Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Ruben,
I am also in Monterrey and have used digium hardware on R2 and PRI.
MFC/R2 is not supported by digium but the zaptel driver requirement is
the same.. what changes is using libpri vs unicall.
Just go ahead and ask them for the firmware update or as Tzafir says
use a newer zaptel that should include the updated firmware.
If in trouble add me to gtalk I'll try to help out any way possible,
Lex
On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
Quote: | On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
|
Generally you can always use a newer zaptel.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
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ruben.zamora at zys.co... Guest
|
Posted: Tue Apr 08, 2008 11:12 am Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Lex
Thanks, I all ready download the last svn branches from zaptel.... And i
am going to test these afternoon.
My phone number es 81-83481611.
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Ruben,
I am also in Monterrey and have used digium hardware on R2 and PRI.
MFC/R2 is not supported by digium but the zaptel driver requirement is
the same.. what changes is using libpri vs unicall.
Just go ahead and ask them for the firmware update or as Tzafir says
use a newer zaptel that should include the updated firmware.
If in trouble add me to gtalk I'll try to help out any way possible,
Lex
On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
Quote: | On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
|
Generally you can always use a newer zaptel.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
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faraz.khan at emergen.biz Guest
|
Posted: Wed Apr 09, 2008 8:54 am Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
The newer zaptel (1.4.10) says it includes firmware 1.16 from the
CHANGELOG:
firmware/Makefile, kernel/wctdm24xxp/base.c,
kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
wctdm24xxp's VPMADT032 firmware to version 1.16
However there seems to be no way to get this firmware and it does not seem to be included. It checks my firmware and says 1.07 is okay.
The URL provided does not contain firmware for the VPMADT032
I* have logged a query with digum. Is there a URL to get this firmware from?
On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks, I all ready download the last svn branches from zaptel.... And i
am going to test these afternoon.
My phone number es 81-83481611.
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Ruben,
I am also in Monterrey and have used digium hardware on R2 and PRI.
MFC/R2 is not supported by digium but the zaptel driver requirement is
the same.. what changes is using libpri vs unicall.
Just go ahead and ask them for the firmware update or as Tzafir says
use a newer zaptel that should include the updated firmware.
If in trouble add me to gtalk I'll try to help out any way possible,
Lex
On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
Quote: | On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
|
Generally you can always use a newer zaptel.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| --
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.111.111.320 x200
www.emergen.biz |
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creslin at digium.com Guest
|
Posted: Wed Apr 09, 2008 1:37 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
|
|
Faraz R. Khan wrote:
Quote: | The newer zaptel (1.4.10) says it includes firmware 1.16 from the
CHANGELOG:
firmware/Makefile, kernel/wctdm24xxp/base.c,
kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
wctdm24xxp's VPMADT032 firmware to version 1.16
However there seems to be no way to get this firmware and it does not seem to be included. It checks my firmware and says 1.07 is okay.
|
We had to back that version of the firmware out due to release related
problems. As for all problems related to the VPMADT032, if you have any
issues, please contact technical support. They will be able to help you
with whatever issue you may have.
Matthew Fredrickson
Quote: |
The URL provided does not contain firmware for the VPMADT032
I* have logged a query with digum. Is there a URL to get this firmware from?
On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks, I all ready download the last svn branches from zaptel.... And i
am going to test these afternoon.
My phone number es 81-83481611.
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Ruben,
I am also in Monterrey and have used digium hardware on R2 and PRI.
MFC/R2 is not supported by digium but the zaptel driver requirement is
the same.. what changes is using libpri vs unicall.
Just go ahead and ask them for the firmware update or as Tzafir says
use a newer zaptel that should include the updated firmware.
If in trouble add me to gtalk I'll try to help out any way possible,
Lex
On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
Quote: | On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
|
Generally you can always use a newer zaptel.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
| _______________________________________________
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Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc. |
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ruben.zamora at zys.co... Guest
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Posted: Wed Apr 09, 2008 9:37 pm Post subject: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032 |
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Today I Install Zaptel 1.4.10 and compiled. No good result.
Then Digium Support send me the last firmware of VPMADT032, and
installed, at the first sight there was no good news.
But then i move in the driver wcte12xp in the file base.c and i have
better results.
Matthew Fredrickson escribi?:
Quote: | Faraz R. Khan wrote:
Quote: | The newer zaptel (1.4.10) says it includes firmware 1.16 from the
CHANGELOG:
firmware/Makefile, kernel/wctdm24xxp/base.c,
kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
wctdm24xxp's VPMADT032 firmware to version 1.16
However there seems to be no way to get this firmware and it does not seem to be included. It checks my firmware and says 1.07 is okay.
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We had to back that version of the firmware out due to release related
problems. As for all problems related to the VPMADT032, if you have any
issues, please contact technical support. They will be able to help you
with whatever issue you may have.
Matthew Fredrickson
Quote: | The URL provided does not contain firmware for the VPMADT032
I* have logged a query with digum. Is there a URL to get this firmware from?
On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks, I all ready download the last svn branches from zaptel.... And i
am going to test these afternoon.
My phone number es 81-83481611.
Thanks
Ruben
Lex Lethol escribi?:
Quote: | Ruben,
I am also in Monterrey and have used digium hardware on R2 and PRI.
MFC/R2 is not supported by digium but the zaptel driver requirement is
the same.. what changes is using libpri vs unicall.
Just go ahead and ask them for the firmware update or as Tzafir says
use a newer zaptel that should include the updated firmware.
If in trouble add me to gtalk I'll try to help out any way possible,
Lex
On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
Quote: | On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
Quote: | Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have UNICALL support.
So... I need to investigate more or wait for a new zaptel or anything else.
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Generally you can always use a newer zaptel.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
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