search for: mediant

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") in new stack -- Called mediant/1 -- SIP/mediant-088a1a18 is circuit-busy == Everyone i...

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) ?=>? PSTN (E1) > ? > ? > I will be very grateful for the help. > ? > Thanks! > > ?...

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: http://lists.digium.com/pipermail/ast...

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 send faxes to the pstn (through E1 PRI) and viceversa... What should we do to make this work properly? what parameters in zapata? mediant 1000? Thanks in advance for all yo...

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:...

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or anythin...

...use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when i press 1627 then it is wait for 5 second and then rining start alternative press '#' what is the method to break this space of waiting after dialing my extention.conf ;North Delhi NOC Extention exten => _16XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _16XX,2,Playback(vm-nobodyavail) exten => _16XX,102,Playback(all-allbusy) ;Mumbai NOC extention exten => _22XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _22XX,2,Playback(vm-nobodyavail) exten => _22XX,102,Playback(all-allbusy) exten => _17XX,1,Dial(SIP/media...

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 ??) and/or what configuration on asterisk side ?? if anyone has it working, please send me the ini file for the mediant and the zaptel.conf etc. would b...

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 => 3911700,1,Dial(SIP/100) When I dial...

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 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 PSTN party answers or the transfer goes to voicemail. This happens regardless...

Dear All I am new in this list right now i am working on asterisk server and deploying asterisk PBX in my organization now i have alread setup Avaya PBX and i want to intergrate my asterisk through mediant 2000 [asterisk]-----[mediant 2k]--------E1-trunk------[Avaya] this is my setup now i want to create dialpan so how to forward call in to existing avaya setup means i have not good knowledge of dialpan routing call is there any configuration example to router call on asterisk this is poss...

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. -------------...

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 wh...

hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely...

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...

...es dentro de una lista que cumplan con condiciones especificas reemplazando los valores antiguos por los nuevos (los valores de la lista están ordenados). Ejemplo: a<-list(c(1,2,2.5,3,3.3,4,4.1,4.7,5),c(1,1.2,2.1,3,3.2,5)) names(a)<-c("a1","a2") a$a1<-a$a1[a$a1>3] Mediante la siguiente función me permite realizar lo anterior rec<-function(indice,valor){ z<-paste0("a$a",indice) b<-paste0(z,"<-",z,"[",z,">",valor,"]") eval(parse(text=b),parent.frame()) } rec(1,3) Lo que necesito es crear una fun...

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 syste...

...nsaje original----- > De: r-help-es-bounces en r-project.org [mailto:r-help-es-bounces en r- > project.org] En nombre de Jorge Tornero - Listas > Enviado el: martes, 03 de diciembre de 2013 13:30 > Para: r-help-es en r-project.org > Asunto: [R-es] seleccionar columnas de un dataframe mediante variables > > Buenos días a todos: > > Para empezar, y como este es mi primer mensaje a la lista, me > presentaré. Mi nombre es Jorge Tornero y trabajo en el Instituto > Español de Oceanografía como Ayudante de Investigación. No soy usuario > habitual de R pero estoy tomando...

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 this type of problem regards Satish patel ---...