search for: kenn10

I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will "glitch" or stutter randomly from time to time. Now, what is interesting is that this happens even with the phone totally disconnected from any

...nk everyone for their hard work and fact finding. I think you have tremendous customer support! It only took a couple of days to track and correct this issue. I'm extremely pleased with your phones and your support! Joe McConnaughey ----- Original Message ----- From: Melissa Lee To: kenn10@comcast.net Sent: Wednesday, August 24, 2005 4:05 PM Subject: FW: Asterisk and MWI Joe Following is the findings from our expert. Please let me know which version of Asterisk that you are using and please also forward your extension.conf to me so that we can compare the files that work...

...Polycoms and Grandstreams out of the water once the Asterisk and other open source communities understand their quality and flexibility. I'd like to be on the cheering squad! Joe McConnaughey ----- Original Message ----- From: "Melissa Lee" <melissa.lee@sayson.com> To: <kenn10@comcast.net> Cc: "David Sayson" <frisketdog@hotmail.com>; <asterisk-users@lists.digium.com> Sent: Monday, August 22, 2005 4:30 PM Subject: RE: MWI problems on 9133i Hi Joe: Thank you for your feedback. MWI problem on 9133i is a known bug and will be fixed in the August...

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again

I agree with Steve. The ability to have bridged extensions appear on multiple phones is a vital to many business users. For Asterisk to seriously take on the Avaya's and Nortels and Cisco's of the world, this is a feature that needs to be implemented. It is apparent that this is not currently supported on Asterisk so I'd hope someone is looking at it for a new feature.

Running Asterisk Head 1.0.9. Below is a trace of a call delivered to my system which had no caller ID. For some reason, dialparties.agi shoves the incoming provider's IP address into the caller ID so you never have a call that is screened for PrivacyDirector. Is anyone else seeing this issue as well? Have I missed a patch? This call shows on the display with a name of "Unknown"

The SC420 does not support APIC (at least not with current BIOS). The Digium TDM400B card seems to always seek the same IRQ so it locks one down. The clone X100P cards often get a shared IRQ, even if vacant ones are available. I've found both the Digium card and the clones very frustrating with this system. Thus the reason Digium does not recommend it. I've gone to a mostly SIP

Tim - My recommendation to you is to get Asterisk@Home. It sets up the Linux and most of the Asterisk for you. Up and running in under two hours. I personally did not want to dig into all the crap in Linux and a lot of the Asterisk setup, so this was a great option for me. Now I'm exploring more about Asterisk without all the pain of having to do it all up-front.

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the 480i

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the

Does the SugarCRM included with AAH 1.3 not meet this criteria for you? ------------------------------ Message: 3 Date: Tue, 26 Jul 2005 17:48:08 +0100 From: "JunkMail" <junkmail@segurajuda.dyndns.org> Subject: [Asterisk-Users] Dial using URI(web) or using FORM(web) To: <asterisk-users@lists.digium.com> Message-ID: <009101c59201$cd5cc9c0$0a00a8c0@segurajuda.local>

Hello - I've been using Broadvoice with Asterisk for a couple of months with no issues. Today, it has stopped registering. The Sip Debug from CLI is below. It tries to register five times and then gives up. Any suggestions? As you might suspect, I have not been able to get Broadvoice on the phone and usually get cut off after being on hold about 5 minutes. Strategic portions of IP

Andi - I have experienced the same issue you mention and gotten no reply as to a way to fix it. I finally implemented "blacklist" into my Asterisk and added "Anonymous", "anonymous", "unknown", "Unknown", etc., into my blacklist file. When those come in with an IP address instead of a phone number but have no real name, they get the

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date: