search for: audiocoding

Displaying 20 results from an estimated 210 matches for "audiocoding".

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... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm
2010 Sep 18
2
Audiocode Median 2000 Gateway with Asterisk ?
Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I
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
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
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,
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
2007 Jul 10
0
Asterisk, AudioCodes, Caller ID
Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up
2006 May 25
1
[asterisk-biz] RE: OT: AudioCodes MP124-C/FSX/AC/SIP
Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP
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
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
2007 Jun 26
1
call fail from audiocode to sip trunk
Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")
2003 Jul 24
2
audiocodes fxs
hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030725/ae4b2f25/attachment.htm
2009 Feb 09
0
Audiocodes - Disconnect Supervision
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 Disconnect and Polarity Reversal settings, to no avail. Anyone experienced this issue with Audiocodes or
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
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:
2004 Jul 08
3
Audiocodes -> Asterisk Implementation
Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found Jul 6 15:12:10
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