similar to: no sound between extensions

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

Hi, Aastra IP Phones have two configuration files on TFTP, aastra.cfg and <mac>.cfg. Both are in text format, which makes editing easy. And aastra.cfghas system wide settings and <mac>.cfg has settings for each indivifual phones. This makes it really easy to change the global parameters system wide by changing only one aastra.cfg file. On the other hand, as I could understand, for

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?

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:

I installed SPA942 and SPA2101, and experimented with TFTP and HTTP provisioning. It all went smooth for many hours. But then all of a sudden it stopped reading configs from both from TFTP and HTTP. Now I am trying to troubleshoot and cant't find the problem. Once in a while, it does read from TFTP and/or HTTP, but then again, stops reading at all. My other phones, i.e. Grandstream and Aastra

Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you.

Hi, I've Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. I've Asterisk@Home server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing

Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... Kim __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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

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, 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

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

Anyone else have problems with phones like SPA-922, SPA-921, etc? Inbound audio is perfect but the remote end reports audio quality issues on the audio the handset is sending out. It's not the network.... I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the least problematic but its still an issue. Doesn't matter if we send the calls to the PSTN

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

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,

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

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

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

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

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