similar to: SIP & H323 gateway

Hi, How can I make calls from Asterisk client to NEC NEAX 2400 traditional phone ? Is it possible to have a connection between Asterisk and NEC NEAX 2400, since NEC-NEAX2400 is an IP-PBX and supports SIP. Please help me to find a solution ;;; Thanks & Regards Ram Kumar Customer Support Engineer Barcode Gulf LLC Dubai , UAE Mobile : + 971 50 5594178 Email : Ramkumar@barcodegulf.net

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line -----> mp108fxo -----> Asterisk ------> mp108fxs -------> fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid

Hello all, Has anybody cofigured in SIP the Audiocodes MP108 FXS in a way that each port is an extension of the Asterisk Box ? So each port can have it's own mailbox, etc ? Regards, Jorge A.

Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it

Linux: RedHat 7.1 Samba: 2.2.7 Windoze #1: 98SE Windoze #2: W2K Here is the situation: copy files from W1 to Linux. At same time, transfer those files from Linux to W2. Of course, the transfer from Linux to W2 doesn't occur until the particular file has completed the transfer from W1 to Linux. This senerio of dual transfer from the same area on the Linux disc will ultimately case the smbd

Then if let say instead of buying TDM400P cards I get this : Clipcomm CG-410 Quad FXO Gateway is it any good? They also sell Quad FXS Gateway. Clipcomm seems to sell the cheapest Quad FXO/FXS Gateways arround so I'm wondering if it's working fine with asterisk. I found this one too but at a lot higher price : AudioCodes MP108 8-Port FXO Analog Gateway (SIP) I need to buy a

Hi everyone: I'm new to the list and forgive me if my question has been discussed million times before. I have an asterisk box up and running supporting about 10 extensions. I can setup extension to forward incoming calls to an external number when its unavailable. My question is that is it possible to have setup like this: Someone calls the number, or getts to an extension, instead of being

Greetings, We found out today that rsync has been missing some files. Here is the senerio: We have two nfs servers in different datacenters, one being the backup to the other. We want to keep them up to date so we use rsync inside crontabs. We have Solaris, AIX, True64 and VMS systems reading and writing to the nfs drives. I am running rsync on Solaris 8 servers. What

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

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS.... Anybody have any ideas? ________________________________________________________________ Sent via the WebMail system at

New to Asterix and perhaps someone can help. The plnned configuration is that the Quintums are to register to the Asterix and the signalling to be handled by the Asterix but the media (G 729 code) to be directed to the service provider. Thanks Shaun

Hi, I have a very frustrating problem and I can't seem to find a solution. I am running redhat 8.0GPL and samba 2.2.8 on two linux boxes. The first linux box is my storage (used by multiple pc's). The second is my web-server running apache and php. The web-server recieves files which are meant to be stored to the first linux box and processed by other computers on the network. The

Does anyone have any success using AudioCodes FXO terminating calls ? Ehsan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/5cfef736/attachment.htm

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was

Author: wyllys Repository: /hg/zfs-crypto/gate Revision: 1b97a96daa581c4f53b4fd8acceb76a27a9fe324 Log message: 6310540 6290437 causes gss_accept_sec_context not to output ret_flags whn no deleg cred; breaks ssh Files: update: usr/src/lib/libgss/g_accept_sec_context.c

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Hi, Here i have a Matrix MyMatrix <- Name Age --------- ------- ANTONY 27 IMRAN 30 RAJ 22 NAHAS 32 GEO 42 and here i have an array with Minimum and Maximum values. MinMaxArray <- data.frame(MIN = 25,MAX=35) MIN MAX ------ -------- 25 35