similar to: Audiocodes MP-11X configuration to work

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?

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 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.0.1 ;Ram size: 32M Flash size:

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

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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090726/643e5b96/attachment.htm

guys, what firmware version of audiocodes mp104 fxs are you using with asterisk? i'm having sip stack problems. ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/50d5f5de/attachment.htm

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. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

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

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

I am currently installing an Audiocodes MP-108 with Asterisk in Singapore. I am able to make outgoing calls but incomming calls just get cut by the MP-108 I can see the corrosponding frontpanel led light p but the call gets cut so I'm very confused about what could be wrong. Could this problem have something to do

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. Time Source Destination Protocol Info 2934 27.221559 10.0.0.110 10.0.0.109 RTP PT=ITU-T G.711 PCMU, SSRC=0x7D08611A, Seq=7662, Time=1805957855 Frame 2934: 214 bytes on wire (1712 bits),

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

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 in the last couple of days. I have three Polycom Soundpoint IP 500 SIP phones, which

Does anyone have configs on the MP-108 FXO to asterisk setup? I'm doing my best with the limited docs, but having very little success. Thanks, -Darren

My experience with an MP-108 was similar. Incredibly complex to setup, and very little help from MFR, or even ABPTECH, the main US reseller. We just couldn't get it working properly. Ended up with a TE110P with an Adit 600 channel bank, which ROCKS. Unbelieveably easy to setup. No echo whatsoever. -Darren

I'm trying to define distinctive rings for lines in this gateway but don't works. Nothing happen when sending a call...the phone doesn't ring.... The same configuration works fine for PAP2NA devices. Adriano Almeida Flickr agora em portugu?s. Voc? clica, todo mundo v?. Saiba mais. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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)

Hi: We're offering some sort of reward to that one who can help us For this site we are using trixbox and Asterisk 1.2 More info off list. Many thanks, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070927/4b058148/attachment-0001.htm

Hello: Is that possible? Do I have to upgrade first to firmware version 6.4? Does firmware version 6.4 exist? Thanks in advance. Regards. -- Usuario Linux Registrado # 342019 --> http://linuxcounter.net/ <-- skype --> luedcortes gtalk --> luedcortes at gmail.com msn --> luedcortes at gmail.com