similar to: FAX over SIP

Hello. I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and i dont get a ring in the caller phone when I dial from a snom200 to the other snom200 or the mp108fxs, I made a debug with ethereal, and I can see a "Ringing" packet being return from the called snom200 or mp108fxs to the asterisk box, but it is not being re-transmitted to the caller snom200. Altough

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 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.

Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an "all destinations are busy" message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of

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 All, Just installed our very first asterisk system, and we love it! I cant believe the different things you can do with it, just great =] My question is: How do I configure my system to play an mp3 file when a caller gets put on hold? Thanks in advance, Steve.

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

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. What is ztcfg for ?, what does it do ?

Hello. I have been going thru the wiki and asterisk related sites and have not been able to find any documentation about how to configure an Adtran TA750 channel bank. The remote disconnect supervision doesn't seem to be working, when the remote caller hangs up asterisk takes up to 30-45 seconds to hangup the call. Can somebody help? Thank's

Hello again Valery, Valery A.Khamenya wrote: > All benefits, what one could obtain from "LLVM supporting multiple CPU > at single host", one might obtaine from "LLVM supporting multiple CPU > at multiple hosts". Isn't that logical? I see more precisely what you mean, but I don't think it is that straightforward to generalise the benefits multiple CPU on

> Interesting email address there :) > On Thu, 2004-01-08 at 01:18, =?koi8-r?Q?=22?=Valery > A.Khamenya=?koi8-r?Q?=22=20?= wrote: unfortunally some email parsers and email clients deny to work correctly with international conventions :( follow this URL for more details: http://www.python.org/doc/current/lib/module-email.Header.html > On the same machine, LLVM definitely needs to

Hey all- Anyone know why the gluster management console can not get the server's disks information? I have set up the gateway and backend , i use the ssh methord to get all information, now i can add servers and delete the current volumes throw the platform ,but i cann't create volume because on the "select bricks" window has no bricks, anyone know why? best regards

Easy one. I have win98 clients that login and connect to the shares fine. My problem is the user home directory, each user can see their folders and can even open them but they cannot save to their home directory. Each users home dir is setup as 0700 (-rwx------), as are the files and subdir's, which I believe is Read, Write, Execute for the owner of the file. What am I missing? Chris.

Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? The antistatic bag had a small label that has E100P written on it, and the card is a bit different than the T100P I already have, Does Digium use the same boards for both cards ? I don't have an E1 link here, can I test the card just by loading the driver and run

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

Anyone can share me with Music Onhold Configuration sample? Thanks in advance for your help, Kang

I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten =>

Where can i get the wine equvilent to directx, and for that matter where is a good place to get .dll equivilents in general. I am trying to install Doom 95, but it searches for directx and wine cann ot locate its directx equiv. so the assumption would be that they didn't install. I am using a wine version written for mandrake 8.0, if that helps. Thx in advance, Joel

Hi All I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to enable /disbale ec on a zap channel per call? Thanks Robb