similar to: Losing registration - ast 1.4.39 and innomedia 6328-2Re

We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant ---------------------------------------- From: "Zeeshan Zakaria" <zishanov at gmail.com> Sent: Wednesday, October 13, 2010 10:41 AM To:

Dear sir, I got trouble on InnoMedia video phone with Asterisk-1.2.4. If InnoMedia video phone as a caller, then the call will be a success, no any problem. The problem happens: If InnoMedia video phone to be a callee, the call can not make successfully. For instance, Caller 23267668 dialed to callee 23267663, callee 23267663 was ringing. If I picked up the callee's phone, then abnormal

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

All, I've migrated a share from an older samba server (3.0.33) running on a Sun Solaris box to a newer samba server (3.6.23) running on a RedHat Linux 6 box. For the most part I've tried to keep the configuration the same (simply to make the migration easier since I'm not an expert on Samba config options). I got the authentication working (security=domain) but now I'm

Greetings, Brian C. Huffman! > I've migrated a share from an older samba server (3.0.33) running on a > Sun Solaris box to a newer samba server (3.6.23) running on a RedHat > Linux 6 box. > For the most part I've tried to keep the configuration the same (simply > to make the migration easier since I'm not an expert on Samba config > options). > I got the

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.39 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

What sort of things should I watch out for when upgrading from 1.4.39 to 1.8.5 Thanks, -- Joseph

Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be

Does anyone know if someone makes a hard video phone for SIP. Dave

Using rsync-2.4.6: Is a times difference supposed to cause a write? Also -t vs -I makes no difference. Below shows the problem, I think: [dmahurin@pc16 /tmp]$ mkdir x y [dmahurin@pc16 /tmp]$ cp /bin/ls x [dmahurin@pc16 /tmp]$ ls -l x/ls -rwxr-xr-x 1 dmahurin users 43024 Nov 13 12:46 x/ls [dmahurin@pc16 /tmp]$ rsync -vrtW x/ y building file list ... done ./ ls ./ wrote 43112 bytes

Has anyone tried SunRocket with Asterisk? http://www.sunrocket.com/ The $199/yr. plan seems like an excellent value, and most reviews have been favorable. However, I don't know if it is possible to obtain the SIP credentials, so one can bypass their "gizmo". Thanks, Stewart

If you have been affected by the SunRocket / ALLO folding issue, ChiliTech would like to extend our hand to you to help you in this time. We will transfer your numbers to us for no cost, and will match your SunRocket or ALLO rate. Please contact us at 1-866-678-6858 x 126 or e-mail mhoppes at chilitech.net. We have been around since 2001 serving the Internet community. Matt Hoppes ChiliTech

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

I have a NY 212 packet8 service if you would like to work with me to set this up on my A@H service, I'm happy to test this with you. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Adam Megacz > Sent: Sunday, 7 August 2005 5:29 PM > To: asterisk-users@lists.digium.com

I'll take either.... Actually now that I have had a chance to think about what I did (sorry bad week here). Yes, I will admit I did patrionize the users list... sorry if I offended anyone. I just figured I'd try to help any SunRocket users out that may not be on the biz list. If you review my history, you'll see I only post business stuff to the biz list. This is an exception.

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: <reginfo> <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.callBack="*97"

Does anyone here implement traffic shaping with shorewall? I need to shape BitTorrent traffic on my network so that upload/downloads do not overwhelm normal function or, even more importantly, my imminent conversion to VOIP for all telephone service. I followed the shorewall documentation guide but am not sure if what I have done is the Right Way Of Doing Things. Nor am I satsified with the

Full_Name: Allan Stokes Version: 2.8.1 OS: XP Submission from: (NULL) (24.108.0.245) I've just spent a hellish six hours trying to create my own R package with a bare bones "hello world" R function inside. I was able to create a package.tar.gz file eventually with much perseverance. My remaining problem is that when I try to install my simple package under RGui, it hangs.