Displaying 20 results from an estimated 1000 matches similar to: "Audiocodes -> Asterisk Implementation"
2004 Apr 23
6
Polycom registration
I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone
registered on an asterisk box but am having no luck. I get the
following errors 192.168.22.196 being the phone and 22.254 being the
asterisk box..
Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request:
Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for
2004 Jun 18
1
app_prepaid NAT issue
I was able to get app_prepaid working, but unfortunately I am getting one
way audio on the phone that I was placing the call from. It is behind NAT.
It appears that the app_prepaid is not taking this into consideration since
I see:
Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route:
Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic>
Jun 18 17:46:25
2006 Oct 22
3
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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2007 Jan 16
4
Audiocodes GPL
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 <Yaniv.Nizan@audiocodes.com> wrote:
>
>
>
> I doubt that we are running Linux on the MP-202. Perhaps there is a
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2003 Oct 01
1
Audiocodes gateway and asterisk
Is anyone on the list using an Audiocodes gateway with asterisk and SIP?
I'm looking at that platform, but I have a couple of issues:
1) Echo cancellation. The echo that I'm hearing with an X100P is
unacceptable. Does the Audiocodes do better?
2) Line signalling. I'm using Kewlstart with the X100P, but it looks like
the audiocodes uses loopstart only. How does this work with
2015 Sep 25
2
Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)
Does anyone have any information for me?
Welinghton.
Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>:
> Hello!
> ?
> I am setting up an Asterisk server with a Mediant 1000 (Audiocodes)
> to make external links. Does anyone have any manual or instructions on
> how to proceed?
> ?
> Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?
2005 Jun 28
1
audiocodes
Is anyone on this list using and audiocodes FXO gateway? I have
Asterisk(1.07 on OS X) setup and working fine, including SIP phones
and IAX2 phones - I can make outbound calls just fine and receive
inbound calls just fine. However, I can't seem to find the right
series of DTMF settings on the AudioCodes to allow DTMF tones to be
sent after an outbound call is connected(phone banking,
2017 Apr 29
2
configure AudioCodes MP-112 with Asterisk.
I've MP-114 that is working configured and working OK with my Asterisk
but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension.
Anything longer than 3-digits is cut off, example I dial extension 1000:
[Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension
2016 Apr 29
1
T.38 with Audiocodes gateway
Hello,
I'm helping a colleague (*) which has the following setup:
ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>--
Audiocodes MP-112 --- <FXO/FXS> --- Fax machine
My issue is the following :
Audiocodes gateway reject INVITEs with 488 Not Acceptable Here
It seems this gateway requires t38 settings to be present in SDP body in
the very first INVITE.
My
2003 Jul 24
2
audiocodes fxs
hi guys,
have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing?
~kelvin
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2009 Dec 28
1
AudioCodes MP-114 making calls via FXO
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
Whenever, I try to call out I get a busy signal.
--
Joseph
2008 Jul 24
2
Audiocodes MP-11X configuration to work with Asterisk
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?
2006 May 24
2
OT: AudioCodes MP124-C/FSX/AC/SIP
Just a question, has anyone knows how to blank or factory reset an
AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit).
I purchased them second-handed with no manuals (thank god for the
internet!!) but i guess the pdf manual I have does not have the
section of factory-reset.
Also, any sucess stories with:
AudioCodes MP124-C/FSX/AC/SIP
2010 Jan 12
1
AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration
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 purchase the unit from they we are just emailing back and forth and the unit is still not working.
Can
2006 Jun 27
1
Voip / AudioCodes MP-108 Help Needed
Hello,
Anyone here have experience with Audiocodes MediaPack MP-108 Gateways?
I would be willing to pay someone for advice and support with configuring my
gateways for a telemarketing project I am starting. My experience is
somewhat limited but all I want to do is make outbound calls just like I
would on normal pots lines. (That's the best way to explain it) I do not
need any special
2009 Dec 31
1
AudioCodes Caller ID
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 <pstn-1270>
[Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite:
2007 Feb 11
1
Debugging a SIP / AudioCodes Problem
I have 2 identical AudioCodes MP-112s. They have the same config except for
the SIP usernames/passwords and the device IP. The configs in extension.conf
and sip.conf are also identical. On one box, when I pick up the phone, I
get a fast busy and the logs/debug show an automatic hangup. On the other
device, I can make calls without a problem. I can even call the phone that
can't make a
2011 Jun 08
1
Asterisk and Audiocodes PRI card
Hello list,
can anyone tell me if this card :
http://www.audiocodes.com/product/ipm-260-sip
is compatible with Asterisk (DAHDI) for use as PCI PRI card ?
Kind regards,
Jonas.
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2007 Jan 15
2
Audiocodes Mediant 1000, Polycom, and no ringback on transfer
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 is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any