similar to: AW: Which IP phones have best voice quality, preferably under $150

Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some

I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, November 01, 2006 11:17 AM To: Asterisk

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Hi, Recently I've noticed on a customer's GXP-2000 phone that it loses its IP addresse for a few seconds, audio goes blank obviously, and after about 30-60 seconds get the same IP addresse back and resumes the call. This shows that call was not dropped but phone lost connection with the server, whereas the caller on the other end was still talking. This is just unacceptable as this is

Hi, 'sip show peers' display info this way Name/username Host Dyn Nat ACL Port Status 229/229 xxx.39.12.58 D N 63969 OK (26 ms) 228/228 xxx.39.12.58 D N 63961 OK (32 ms) But this makes is difficult to understand which extension belongs to whom, and I want to set it so I can see it like this:

For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas,

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]

On one of my servers I have many IP phones connected, locally and remotely, and all of them have register expiry set for 1 min. But on Asterisk CLI I see only one of them, extension 502, re-registering every single minute. None of the others re-register or info doesn't come on the CLI screen. This one phone is remotely connected. What is the reason for this. And also everytime it re-register,

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of

Hi everybody, One of my customers wants to buy IP Phones and Asterisk solution, but his requirement is if he'll not be happy with Asterisk, his phones should be able to work with other IP PBX systems as well, so that he doesn't have to buy new phones again. After all IP Phones is the main investment. He'll most probably go with Nortel IP PBX system if he'll be not satisfied with

Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read

Greetings, How can I call cheap to UK cell phones. I am located in Toronto, Canada, but need to call UK cell phones both from Toronto and London. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080226/480f7a28/attachment.htm

Hi, Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

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:

Greetings, Sounds like a simple thing to do, but I was not able to do it on these particular phones. Snom wiki was not helpful. My client wants to keep his phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours difference. The phones are provisioned from a tftp server. If I remove 'dst' value from the provisioning file, on bootup phones force users to pickup

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve