search for: mp108

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...

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 * co...

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

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.

...DM400P 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 conference phone also and I'm looking at Cisco or Polycom. Anyone tested one of the 2? Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a rando...

...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'm not mistaken Asterisk can entertain both H323 and SIP so I need to configure Asterisk as SIP and H323 gatekeeper to take calls and route to our International Carrier, I installed Asterisk on Fedora core...

...e 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 with this being a Singapore Phone system and MP108 designed to work only with american phone systems. any help will be apreciated. thanks -- regards Vikram

...at extension has no callwaiting, no callforward, no voicemail, and hangs up the call inmediately with a "nobody is available to take your call right now" message, making the queue useless. My PSTN connection is an AS5300 in SIP, my extensions are analog phones connected to an Audiocodes MP108-FXS with SIP. This is the output from CLI with High Verbosity: XXX.XXX.XXX.XXX is the IP of the AS5300, 8521 and 8522 are the only two agents in the queue that have inbound calls in progress when a third call arrives and this happens. 8500 is the queue number -- Executing SetVar("SIP/...

I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be

...o head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable.. registrations were ok, etc.. but now in head it's borked. verbose = 30 debug = 30 sip debug on.. *CLI> <-- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 Max...