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

Displaying 20 results from an estimated 5000 matches similar to: "Which IP phones have best voice quality, preferably under $150"

2006 Oct 30
6
How to do Automatic Daylight Saving on Grandstream GXP-2000
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
2007 May 18
5
Phone losing IP address for a few seconds but doesn't drop call
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
2006 Nov 01
0
[SPAM HEADER] - Which IP phones have best voice quality, preferably under $150 - Email found in subject
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
2006 Nov 01
0
AW: Which IP phones have best voice quality, preferably under $150
snom 300 :"> CS -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Kristian Kielhofner Gesendet: Mittwoch, 1. November 2006 12:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Which IP phones have best voice quality,preferably under $150 Zeeshan Zakaria
2007 May 16
1
How to remote reboot Grandstream GXP-2000
I occasionally have to remotely configure and reboot customers' grandstream phones. I use tftp, so make changes in the tftp config files, but then need the customers manually reboot the phones. How can I reboot the phones myself remotely. -- Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Mar 09
1
Which VoIP router and switch to use for medium size business
Hi everybody, What is a proper setup for a medium size business with about 20 IP phones and 20 computers. Right now they are using a regular Linksys router which we use at homes. Their switch is also a very standard switch. Now they need to put there something better and VoIP compatible. What people use out there in serious and professional VoIP installations for medium size businesses? Is there
2010 Aug 24
9
Should I move to 1.6 or 1.8, or stay with 1.4?
Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring
2010 Oct 20
4
Recommendation for a new server
Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm
2010 Nov 01
4
FW: Under heavy attack
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
2006 Dec 10
10
Recommendations for QoS, PoE Switches
Hi all, For a top quality setup, I will need to install high quality VoIP switches with QoS and PoE. My potential customer should not have any problem with call quality. Experienced folks, Please advice me what switches to install and at what price. I may need it for upto 100 phones. What else should I consider so that phones work without problem along with the computers on the same network?
2008 Oct 17
5
How to add contexts in asterisk realtime?
Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for
2009 Jul 11
2
Suggestions for web based soft phones
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,
2010 Oct 23
3
Cepstral voice quality not good
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
2010 Oct 30
8
Under heavy attack
My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part
2009 Jul 14
3
Why CDR is recording dst value = h?
For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to
2006 Jan 30
5
Grandstream Budgetone mass deployment?
Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working?
2006 Oct 31
3
Asterisk and ARI (Aterisk Recording Interface) integration problem
Anybody knows why ARI gives this error message when I enter extension number and password. *Warning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to open stream: Permission denied in * /var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only
2010 Jun 08
6
reloading realtime sip peers
Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip
2010 Mar 18
6
Asterisk DIES with no trace. PLEASE
Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >