similar to: asterisk + mediant 2000

Dear All I want to integrate asterisk with mediant so anybody have configuration for this setup [asterisk]----------[mediant]------[avaya] this is my setup so what is the basic configuration for this setup --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part -------------- An

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when

Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3

Thanks James, What i need is to make the fax machines connected to the audiocodes mediant 1000 be able to send and receive fax throught Asterisk (connected to a pri) I know it's not reliable, but it should work at leaste, what should i do on Asterisk and Mediant to make this work? Im quite confuse with all these fax issues :S Thanks in advance > > Message: 11 > Date: Fri, 9 Apr

Hello my friends, I want to make fax work in the following scenario: My versions are: Asterisk 1.4.21.2 WANPIPE Release: 3.4.7 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P The E1 pri is connected to our Sangoma A102DE, we also have a SIP Mediant Audiocodes 1000 where we have some fax machines connected to fxs ports, what we need is to make fax machines through mediant

Hi, I have been using/working on asterisk for some time now and presently was trying to configure asterisk to work with digium cards. It worked fine with the fxo/fxs cards, but now i'm trying to get it working by interfacing it with mediant t1 port. no avail ....... anyone out there got it working, what particular configuration used on mediant (isdn signalling, framing, coding etc ??)

Hi, I am configuring an audiocodes Medant1000 to talk to my asterisk box. So far I have successfull in landing a single call from mediant to my *box. my sip conf is as follows: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3911700] type=friend host=dynamic dtmfmode=info secret=blah context=sip where 3911700 is my E1 telephone no. in my extensions.conf I have exten =>

Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog phone Regards

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) ?=>?

Hi, This is my first experience with a mediant 1000 and an Asterisk Trixbox. the mediant has 12 FXOs and 12 FXSs, and I want to use it them all. I will have extensions connected to the FXS ports, and lines to the FXO. Can anyone guide me, please? regards, -- Guillermo Garron "Linux IS user friendly... It's just selective about who its friends are." (Using Ubuntu, Debian,

I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi FOlks, I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar

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

Greetings... I'm working on getting an AudioCodes Mediant 2000 big box o' PRI's going with Asterisk, and am running into a problem with DTMF handling. The box is sending DTMF packets to Asterisk as INFO packets, and they are apparently being seen by Asterisk. However, the DTMF knowledge doesn't seem to actually do anything -- the VM system doesn't recognize the digits,

Before an IP Phones can be registered to an Asterisk server, the extension for it must be configured in Asterisk. Usually, Asterisk adminintor must add the extension by hand. Is there any library, API to do this by software??? For example, i want to develope a software that add new extensions to Asterisk system, sothat, any IP Phones can use that extensions to establish a call. I'm digging

Dear all I have setup with mediant 2000 with avaya now i want to install E1/PRI card with asterisk and trunk with E1 with Avaya E1 port so i want to buy E1 card for asterisk so which card is best and cast effective for my setup i want 1 port E1 card so can you suggest me which card is best for my setup and i want QSIG singaling with avaya Regards satish patel

What does the following EDAC problem means? The machine is a AMD 64bit box running Centos 5. It looks like some problems aroung AMD DRAM Memory controller. But what does it really mean b/c most of my AMD boxes has these messages in /var/log/messages. Please help. ... Aug 1 23:29:40 ccn128 kernel: EDAC MC: Ver: 2.0.1 Jun 12 2007 Aug 1 23:29:40 ccn128 kernel: EDAC MC0: Giving out device to