similar to: Audiocodes MP-114 noise

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

>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

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

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

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?

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

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using

We must have had the magic version of 1.6.x then, because we increased our buddy watch limit from 8 to 48 in that version. -----Original Message----- From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Wednesday, October 11, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Limit

Hello Folks, I'm trying to decide here between a few cards for connecting an Asterisk box to a single E1 channel (either PRI or R2 signaling): - Digium E100P: has been replaced by the TE110P below, but can still be had at places like digitnetworks.com for $475, and I guess there's always a place for good-olde-obsolete cards in the world as long as they work :-) - Digium TE110P:

hi any chance of making asterisk support these? http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032 -38356249088.htm roy

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 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

I have an Audiocodes MP-114 and modem dial-out can not establish connection. Snooping with Wireshark I get a lot of entries like this: No. Time Source Destination Protocol Info 2934 27.221559 10.0.0.110 10.0.0.109 RTP PT=ITU-T G.711 PCMU, SSRC=0x7D08611A, Seq=7662, Time=1805957855 Frame 2934: 214 bytes on wire (1712 bits),

Hello: Is that possible? Do I have to upgrade first to firmware version 6.4? Does firmware version 6.4 exist? Thanks in advance. Regards. -- Usuario Linux Registrado # 342019 --> http://linuxcounter.net/ <-- skype --> luedcortes gtalk --> luedcortes at gmail.com msn --> luedcortes at gmail.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