similar to: Audiocodes MP-20x

It's noisy while talking. Any idea? Thanks in advance. Jason ____________________________________________________________________________________ Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)

I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension

I was able to setup AudioCodes MP-114 to rote calls form FOX to Asterisk and make internal calls: Routing Tables -> Tel to IP Routing: *, *, 10.0.0.109 (my asterisk IP) But I'm not sure how to setup AuioCodes to make calls out via FXO? In extensions.conf [Globals] pstn-5665=10.0.0.157 Whenever, I try to call out I get a busy signal. -- Joseph

I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help?

I have AudioCodes MP-114 and I'm trying to configure SAS (Stand Alone Survivability); when Asterisk is down the MediaPack gateway should forward the call IN/OUT through the gateway (without asterisk in the middle), but it is not working. I'm working with tech. support from the source I purchase the unit from they we are just emailing back and forth and the unit is still not working. Can

Hello, Anyone here have experience with Audiocodes MediaPack MP-108 Gateways? I would be willing to pay someone for advice and support with configuring my gateways for a telemarketing project I am starting. My experience is somewhat limited but all I want to do is make outbound calls just like I would on normal pots lines. (That's the best way to explain it) I do not need any special

Greetings; We are trying to get Asterisk up and happy at our site-we tried VOIP using Sphere about a year ago, spent a *boodle* on expensive hardware and services from a local "expert", but it never was happy. I'm brand-spanking new at VOIP, and I've learned a *ton* getting Asterisk breathing in the last couple of days. I have three Polycom Soundpoint IP 500 SIP phones, which

Hi list, I'm trying to get ready the MP-104 FXO to use qith my box, but when I send calls I hear only dial tone and after a few seconds I get busy signal. I very appreciate your advices. Command line results and SIPconfigurations follows: *CLI>* -- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918", "beep") in new stack -- <SIP/101-09dd8918>

>From atacomm.... ________________________________ From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Thursday, May 12, 2005 2:24 PM To: Wiley Siler Subject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either in June or July. Right now they developers are cleaning up the echo cancellation

Has anyone used the Polycom HDvoice phone yet? I am curious if it uses a different codec. Does it actually sound any better?

I am having the same issue as below. Has this issue been solved or does anyone know an answer? This error recently began and we have multiple phones out of commission. PLEASE HELP!! http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -----Original Message-----

A client of mine has a Snom 320. Usually when he comes in each morning, it is asking him for a password. A power cycle brings it back to normal operation. How do I troubleshoot this further? --Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070221/0e4d3ab8/attachment.htm

Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Snom and the company representing Snom in the US. So

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas?

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An HTML attachment

Does anyone have an actual delivery date on the new Polycom HD IP650's? I'm getting sick of not having a backlit screen and thinking of upgrading. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061018/9937184f/attachment.htm

Hi, Where should I go to get the Polycom`s latest official (non-beta) version? I am registered on the Polycom customer website but that doesn't seem accessible. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/284b67a7/attachment.htm

Hi all, Has anyone had any experience getting Polycom phones repaired? The screen on one of our IP600s got smashed, and I'm wondering if it's worth the effort to get it repaired, or if it'd just be cheaper to buy a new phone. Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd

I think that limit was increased in 1.6.6 or 1.6.7. -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 10/10/2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list On 10/10/06, Eric ManxPower Wieling <eric@fnords.org> wrote: > What I