Does anyone have experience setting up an AudoCodes MP-X with an asterisk
(FreePBX based) system? I would be willing to pay a reasonable amount for
assistance with the MP-X device. I have remote access setup, so no one should
have to leave their comfy chair..
Thanks
David
--
Funny you should ask! I have an MP-202 in front of me right now that I'm
working on. When I get it working, I'll let you know. In the mean time,
what symptoms are you getting?
Mike Diehl.
On Mon, Jul 1, 2013 at 4:07 PM, David Wessell da...@ringfree.biz wrote:
Does anyone have experience
Slightly off topic but I have an Asterisk server with Audiocodes HD310
phones, they are running 1.2.2 software and I have 1.6.2 - they come as img
files , but I cannot work out how to load the img file into the phones -
any one know?
--
Looking for help with an initial config of a Mediant 1000 with single
T1/PRI. Need to route calls to an Asterisk server as well as a fax
server. Please email me offlist.
Thanks
--
_
-- Bandwidth and Colocation Provided by
I have an Audiocodes MP-114 and modem dial-out can not establish connection.
Snooping with Wireshark I get a lot of entries like this:
No. TimeSourceDestination Protocol Info
2934 27.221559 10.0.0.11010.0.0.109RTP PT=ITU-T
On 10/14/10 15:38, Bryant Zimmerman wrote:
For which device models?
From: Mark Murawski markm-li...@intellasoft.net
Sent: Thursday, October 14, 2010 3:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Audiocodes firmware
Does anyone
On Thu, Oct 14, 2010 at 11:46 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Crazy. What do you plan on using for an ATA now?
The problems I'm having are getting 500 Server Internal Error on just
about every other call placed out of this mp-118. The box has been
installed and in use
October 14, 2010 3:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Audiocodes firmware
Does anyone have links to the
most recent audiocode
On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Does anyone have links to the most recent audiocodes firmware?
Why not contact Audiocodes?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box.
The problem is, the vendor that sold me the box is really hard to deal
with and has been brushing me off all week on getting firmware.
On 10/14/2010 05:14 PM, Paul
From: Paul Belanger paul.belan...@polybeacon.com
Sent: Thursday, October 14, 2010 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Thursday, October 14, 2010 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box.
That is ridiculous, how hard is it to provide a download link and
disclaimer about no
Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box
On Oct 14, 2010, at 5:40 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On Thu, Oct 14, 2010 at 5:27 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Because audiocodes does not provide support to end users and will tell
you to contact your vendor that sold you the box.
That is
an...@polybeacon.com
Sent: Thursday, October 14, 2010 6:43 PM
To: "Asterisk Users Mailing List - Non-Commercial
Discussion" asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010 at 5:27 PM,
Does anybody knows how to enable Stutter Tone on Audiocodes MP-114?
Similar feature like Sipura has, when message is left in a mailbox the unit
send a 0.5sec ring tone to the phone every 30min. or so.
IN:
SIP Advanced Parameters -- Supplementary Services:
Message Waiting Indication (MWI)
SOLVED!
Correct me anybody if I'm wrong but I think SAS option is for WAN only not for
the case if AudioCodes MP and Asterisk are on the same network.
I was trying to configure the fail-over mode in scenarios:
- Asterisk sever goes down (doesn't happen very often, never happened to me but
it
I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone
Survivability); when Asterisk is down the MediaPack gateway should forward the
call
IN/OUT through the gateway (without asterisk in the middle), but it is not
working.
I'm working with tech. support from the source I
I have AudioCodes 2xFXO / 2xFXS but can not make the FXO port to work
correctly; I can dial out on one FXO port or the other FXO, but not on both.
It depends on Sorting in : Hunt Group Setting (Ascending, Descending)
If setting is set to a Cyclic Ascending I can dial out on FXO port every
second
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch,
have pstn-5665, digest has
Joseph wrote:
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username
mismatch, have
On 12/31/09 13:06, Kevin P. Fleming wrote:
Joseph wrote:
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553
On 12/31/09 13:06, Kevin P. Fleming wrote:
Joseph wrote:
I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO)
AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to
interpret it as authentication:
[Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553
When I configured AudioCodes MP-114 to MWI it keeps complaining bout
subscription without mailbox:
chan_sip.c:15450 handle_request_subscribe: Received SIP subscribe for peer
without mailbox: pstn-5665
chan_sip.c:15450 handle_request_subscribe: Received SIP subscribe for peer
without mailbox:
I solved the problem with calls out via FXO but internal call to to phone
connected to FXS on AudioCodes is not working:
app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause
20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing
On 12/28/09 01:14, Joseph wrote:
I solved the problem with calls out via FXO but internal call to to phone
connected to FXS on AudioCodes is not working:
app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause
20 - Unknown)
== Everyone is busy/congested at this time
The web interface is a bit confusing at first. Here are some of the
steps that I remember off hand. Change as little as possible, makes
it easier to troubleshoot later.
Get the latest code from your vendor (5.6 is what I run)
Configure the proxy to register with
Configuration - Protocol
What what everybody says, it is a good hardware but configuration samples are
not easy to find and going through 500page manual is not easy.
What they are missing is short configuration guide with samples for specific
software like asterisk.
My software version is 5.40A I see early next week
The best document is the two page quick start guide that came in the
box. You want 5.6, and 5.8 should be out soon if you are an early
adopter.
-Jonathan
Sent from a mobile device.
On Dec 27, 2009, at 9:02 AM, Joseph syscon...@gmail.com wrote:
What what everybody says, it is a good
I've emailed my provider Scansouce Communication to provide me with the newer
firmware version but for now I'm trying to setup so I can at least register it
with asterisk.
Can you email me your INI file, I'll compare it to mine.
I'm trying to register both ports with FXO and FXS with Asterisk.
On 12/27/09 09:58, Jonathan Thurman wrote:
The best document is the two page quick start guide that came in the
box. You want 5.6, and 5.8 should be out soon if you are an early
adopter.
-Jonathan
The two page quick start does not work. I've follow the instruction in there,
it is simple and
On 12/27/09 12:10, Joseph wrote:
On 12/27/09 09:58, Jonathan Thurman wrote:
The best document is the two page quick start guide that came in the
box. You want 5.6, and 5.8 should be out soon if you are an early
adopter.
-Jonathan
The two page quick start does not work. I've follow the
I was able to register only one FXO port by setting in:
Protocol Definition - Proxy Registration:
Proxy Name: 10.0.0.109
Registrar IP Address: 10.0.0.109
User Name: pstn-5665
Password: 147
But how do I register other FXO and tow FXS ports?
When I trying to enter in:
Endpoint Settings -
I was able to setup AudioCodes MP-114 to rote calls form FOX to Asterisk and
make internal calls:
Routing Tables - Tel to IP Routing:
*, *, 10.0.0.109 (my asterisk IP)
But I'm not sure how to setup AuioCodes to make calls out via FXO?
In extensions.conf
[Globals]
pstn-5665=10.0.0.157
I have AudioCodes MP-2FXO/2FXS but have a problem registering it with Asterisk.
Any links or pointers to configuration how it is done?
--
Joseph
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does
anyone have a setup file they can share to help me work this out.
Instructions or a link I can follow - thanks.
___
-- Bandwidth and Colocation Provided by
I have an Audiocodes MP-118FXO in production. When an outbound call is made and
the remote party hangs up, the Audiocodes hangs up the call immediately. But if
an incoming call is received and the remote party hangs up, the Audiocodes does
not hang up immediately.
I have tinkered with Current
Sir,
Here is the working Audiocodes MP-11X FXO configurations to work with
Asterisk.
;**
;** Ini File **
;**
;Board: MP-118 FXO
;Serial Number: 905371
;Slot Number: 1
;Software Version: 5.00A.024
;DSP Software Version: 204IM = 209.13
;Board IP Address: 192.168.0.195
On Tue, Dec 30, 2008 at 00:25, Jeff LaCoursiere j...@jeff.net wrote:
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not
2008/12/30 Jeff LaCoursiere wrote:
Oops - I take it back: http://www.audiocodes.com/gpl-lgpl
Looks like they are at least attempting to comply... did you follow these
steps?
Thank goodness you posted that, it's saved weeks of self flagellation after
reading the earlier posts. ;o)
So now we
On 30 Dec 2008, at 09:35, Razza wrote:
2008/12/30 Jeff LaCoursiere wrote:
Oops - I take it back: http://www.audiocodes.com/gpl-lgpl
Looks like they are at least attempting to comply... did you follow
these
steps?
Thank goodness you posted that, it's saved weeks of self
On 28 Dec 2008, at 18:36, Razza wrote:
Please see below Console Messages, Pertinent section of SIP.CONF and
AudioCodes Config.
Console Messages:
Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808
handle_request_register: Registration from 'sip:2...@192.168.10.4'
failed for '192.168.10.4'
AudioCodes blatantly violates the terms of the GPL by not distributing
the source code even after requesting it. Please don't use their
hardware.
On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski ft...@mindspring.com wrote:
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It
What does Audiocodes release under GPL?
j
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
AudioCodes blatantly violates the terms of the GPL by not distributing
the source code even after requesting it. Please don't use their
hardware.
On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not and
later they admitted it did, but refused to supply the source code.
Oddly enough, the Linux distribution is OpenRG,
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not and
later they admitted it did, but refused to supply the source code.
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not and
later they admitted it did, but refused to supply the source code.
On 27 Dec 2008, at 22:54, Razza wrote:
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk
also, I want all the channels to register with asterisk.
I have the FXS channels working fine, I cant acheive that with the
FXO channels, does anyone have any advice or possibly sample
Please see below Console Messages, Pertinent section of SIP.CONF and
AudioCodes Config.
*Console Messages:*
Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 handle_request_register:
Registration from 'sip:2...@192.168.10.4 sip%3a...@192.168.10.4' failed
for '192.168.10.4'
Dec 28 18:14:45
Razza,
I have a MP114 FXO/FXS that I have never got to work , even as an FXS,
even though I have several other FXS's that work fine ie Linksys PAP2
etc.. would you put up your config?
PDE
___
-- Bandwidth and Colocation Provided by
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk also, I
want all the channels to register with asterisk.
I have the FXS channels working fine, I cant acheive that with the FXO
channels, does anyone have any advice or possibly sample configs.
Thanks in advance :)
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It
registers fine and I can call between the MP-114 and other extensions,
but I'm not having much luck with the FXO ports. syslog shows the
problem to be in the MP-114 configuration.
Can anyone help?
, including any attachments. Our mailing address is
454 Sonwil Drive, Buffalo, NY 14225 USA.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frank
Tarczynski
Sent: Thursday, July 24, 2008 8:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
Afternoon All,
Does anyone here have any experience with an Audiocodes Mediant 2000?
I know its a bit 'non asterisk' but i figured you guys are as likely
as any to have come across them. I'm having a few problems with one,
i.e. its not sending screening/privacy flags although it is sending
Hello,
Has anybody seen that Audiocodes gateway is replying with 486 Busy
here when it's actually not (last call ended ~15 seconds ago).
I see this quite often. From other logs i see that previous call ends
at 11:13:01, then app_queue tries to dial at 11:13:14 and fails
numerous times, before
Hi,
I am using OpenSer + Asterisk. I am using a Audiocode MP112 over a
satellite link. The ping time to the server is about 700ms. When
connecting to another carrier there is no delay what so ever. When I
connect it to my test server there is a 3 second delay. From what I
heard my test carrier is
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 27, 2007 10:51 AM
Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk
Sent it to AudioCodes (in a text file). I will let you guys know what the
issue was.
- Original Message -
From: Shanon Swafford [EMAIL
: Re: [asterisk-users] AudioCodes Gateway and Asterisk
When you see [ERROR] in the Message Log, either the MP firmware is buggy
or the far end is sending something out of spec in the SIP Message.
You'll need to upgrade to the latest MP firmware then report this to
whomever you bought it from
://www.abptech.com/support/qa/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Sunday, June 24, 2007 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk
- Original Message
- Original Message -
From: Atis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 21, 2007 5:58 PM
Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk
On 6/21/07, Dovid B [EMAIL PROTECTED
- Original Message -
From: Shanon Swafford [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, June 21, 2007 6:27 PM
Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk
On 6/21/07, Dovid B [EMAIL
Hi List,
I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I
keep getting an error from asterisk of -- Got SIP response 415 Unsupported
Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone
have a hint as to what it may be ?
Thanks.
On 6/21/07, Dovid B [EMAIL PROTECTED] wrote:
Hi List,
I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I
keep getting an error from asterisk of -- Got SIP response 415 Unsupported
Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729.
Anyone have a
On 6/21/07, Dovid B [EMAIL PROTECTED] wrote:
Hi List,
I am trying to call from my asterisk box (1.2.18) to and audiocodes
MP114. I
keep getting an error from asterisk of -- Got SIP response 415
Unsupported
Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729.
Anyone have a
Greetings;
We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle* on expensive hardware
and services from a local expert, but it never was happy.
I'm brand-spanking new at VOIP, and I've learned a *ton* getting
Asterisk breathing
1) dont use MGCP -- SIP is better supported
2) Don't use Audiocodes, they blatantly ignore the GPL license.
On 4/19/07, J. David Bavousett [EMAIL PROTECTED] wrote:
Greetings;
We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle*
: [asterisk-users] AudioCodes MP-104 MGCP?
1) dont use MGCP -- SIP is better supported
2) Don't use Audiocodes, they blatantly ignore the GPL license.
On 4/19/07, J. David Bavousett [EMAIL PROTECTED] wrote:
Greetings;
We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere
AudioCodes is known to violate the GPL and not care at all about it.
On 12/13/06, Mike Clark [EMAIL PROTECTED] wrote:
Anyone have any experience with the Audiocodes MediaPack MP-118? We are
looking at options for a location that wishes to maintain 6 - 8 existing
analog phones in add
I finally have the solution, so thought I would post back to the list for
completeness.
It ended up being a series of changes. First, on the gateway, set Disconnect
on Broken Connection to false. Then, for the Polycom phones, set
voIpProt.SIP.allowTransferOnProceeding to 1 in the sip.cfg.
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
However my contact at Audiocodes claims otherwise
On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote:
I doubt that
Andrew Joakimsen wrote:
2) What does one go about doing to correct GPL violations? Perhaps
someone has a generic legal letter that can be used in these
situations?
Only a copyright holder whose code is being used outside the terms of
the GPL can pursue action against the violator.
Andrew Joakimsen wrote:
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
Googling turns up:
http://www.jungo.com/openrg/openrg.html
OpenRG is a Linux based device
Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon:
Andrew Joakimsen wrote:
I have some Audiocodes units which appear to be running Linux,
according to the unit's own System Log
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
Googling turns up:
On Mon, 2007-01-15 at 15:26 -0600, David Gomillion wrote:
I don't think you can do that. Here's why: on the Polycom's, the
Transfer button doesn't reappear until the transferree picks up the
phone. Unless something changed in the firmware recently. But, if you're
completing it before the 3rd party
Of Anselm Martin
Hoffmeister
Sent: Tuesday, January 16, 2007 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Audiocodes GPL
Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon:
Andrew Joakimsen wrote:
I have some Audiocodes units which
I just put in a Audiocodes Mediant 1000, which seems to be working well except
for one annoyance. I am using Polycom 501's and 601',s and if I do a
supervised transfer of a PSTN call where I complete the transfer before the 3rd
party has answered, the PSTN party hears dead air until the call
On 1/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I just put in a Audiocodes Mediant 1000, which seems to be working well
except for one annoyance.
I don't have any experience with an Audiocodes Meidant 1000, but I'll try to
help you
I am using Polycom 501's and 601',s
We have a
Anyone have any experience with the Audiocodes MediaPack MP-118? We are
looking at options for a location that wishes to maintain 6 - 8 existing
analog phones in addition to a number of new Polycom phones.
Thanks,
Mike Clark
___
--Bandwidth and
Hi all,
Has anybody a clue how to pass the call immidiatly ?
At the MP104 I put at the autodial a number of a sip extention.
Also, I noticed that after those 3 rings the extention is ringing but
not passing the call to voicemail as should be after a while like it
happens when you call from other
Jessee,
Thank you for your help.
I downloaded firmware and sample configuration files.
But the firmware was old version for MP118 and MP124.
Where can i download recent one?
Can i upload only ini file to change
countrycoefficient ?
Regards,
Jason.
--- Jessee J Holmes [EMAIL PROTECTED] wrote:
Jason,I think it's something only supported in the newer firmware. Get in contact with the place you bought the unit from, they should be able to get the latest firmware for you.MP118_SIP_F4.80A.034.004.cmp should work for MP114, this is what we used and all that we could get from Audiocodes, I
Jason,First, before you start reading, get to the latest firmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), there have been significant echo improvements in this version.After many days of working with Audiocodes on this problem and much time spent here by multiple technicians trying to
Jason,Funny enough, I'm working on this exact same problem right now with one of our customers and we're scheduling a conference call with Audiocodes today. I'll get back to you on what I found out. However, in the mean time, you may want to take it up with the place you bought the unit from, they
Jessee,
I tried many combinations of Voice Volume, Input
Gain and packetization time , but it's noisy steel.
I'm using G.711A-law and packetization time is 20ms.
It can be impedance mismatch problem but i cannot
adjust impedance of FXO port of MP-114.
--- Jessee J Holmes [EMAIL PROTECTED] wrote:
Jason,There are a couple things we can try to fix your problem.Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmpLet's try some quick things first though:In your web interface, go to advanced config - channel settings / voice settingsThere are some
Dear Jason,Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well?This start happening immediately, or was the box working before and the problem just started?Also, a quick heads up, make sure before even beginning to troubleshoot an
Thank you Jessee,
Firmware seems to be recent(4.80A.025.004).
For 'noisy', I mean IP Phone -- * -- MP-114 side.
Audio quality of MP-114 -- PSTN -- Analog phone is
good.
I think it can be power ground or gain problem.
Any experience?
Thanks,
Jason
--- Jessee J Holmes [EMAIL PROTECTED] wrote:
Of Andrew Joakimsen Sent: Monday, October 23, 2006 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes MP-20x
Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem
It's noisy while talking.
Any idea?
Thanks in advance.
Jason
Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates
(http://voice.yahoo.com)
___
Sent: Monday, October 23, 2006
1:47 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users]
Audiocodes MP-20x
Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem to find anyone
Subject: [asterisk-users] Audiocodes MP-20x
Has anyone used the AudioCodes MP-20x?
http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem to find anyone actually using
them...
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Hi Has anyone used the AudioCodes MP-20x?I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive.
RegardsAndrew
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On 10/23/06, Andrew Nowrot [EMAIL PROTECTED] wrote:
I've been testing this for 3 weeks now. No problems so far. This gateway has
many features including IPSec and is not that expensive.
Appreciate if you can post the sample configs to wiki or to the list.
There is no information about
The reason not many people have this product, is because this product is not going to be available to the public at this time.Audiocodes will only provide this product (currently due to ship in December) as 1,000-piece minimum orders for the MP-202. The MP-201 will be available sometime quarter 1
If you dont mind me asking a few questions, I am wondering, to what extent have you tested the units? Do all the basic functions (call id, call waiting, call transfer, forwarding, etc) work on the unit? How well do the router functions work? Overall quality and impressions?
On 10/23/06, Andrew
Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdfSeems like a good device, but I can't seem to find anyone actually using them...
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On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Has anyone used the AudioCodes MP-20x?
http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem to find anyone actually using
them...
I am using an AudioCodes Mediant1000 and now
On 9/23/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, Sep 22, 2006 at 11:46:53PM +0200, Morten Isaksen wrote:I only know of the Audiocodes MP124 and I can recommend this. They
are very stable and have some excellent management options. We have200+ of these in service so the management is a
I only know of the Audiocodes MP124 and I can recommend this. They are very stable and have some excellent management options. We have 200+ ofthese in service so the management is a big issue for us.
The downsides. The box itself feels a bit too much plastic like and I would prefer if the put
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