similar to: Sidetone with Snom 370

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

Hi - I've got a new install with a Sangoma A200 and a few GXP2000's. When users are talking over the Sangoma, they get a lot of sidetone (local echo). Internal calls are fine. Where do I adjust that? I assume its in zapata.conf somewhere? thanks Todd

Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the "Recording" button on this phone. The extension I configured on this phone has the values "Recording on demand", an the voicemail enabled. I am using FreePBX to manage my PBX. How should I configure the "Function

We've got our Polycom phones auto-answering for paging. Is it possible to configure a PAP2 to auto-answer for either paging or intercom? If so, how?

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

I am using GS 101 and as I am new to Ip phone arena. I am finding it a bit annoying to hear sidetone, especially when both parties are talking over each other occassionally. In that case, I cannot hear the other party's conversation. Is there any way to suppress it? Is it only GS or it applies to more expensive phone eg Cisco 7960 as well? -- David Kwok Iaxtel/FWD # 17001813482 ext

Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details

Thank you. While I'm currently in the process of migrating to AlmaLinux 8 via an in-place upgrade, I initially wanted to update the base packages on CentOS 7 before proceeding. However, given the circumstances, it seems deploying a separate AlmaLinux machine for testing might be the most efficient approach to troubleshooting the issue. If the problem persists on a fresh AlmaLinux

Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones

Hey I want to generate a discrete uniform distribution as follows: For example: I want to get 278734 records each with a numbers between 7-10. And the sum of numbers in 278734 records to be equal to 2253712. Once this is done, I want to get that printed to an excel file such that Record Value 1 9 2 7 so on 278734 8 The

On 16 Nov 2001, Zaheer Merali wrote: > That is an idea that is coming up in the next ZStreamCaster. > ZStreamcaster 0.1 currently allows you to save a stream to disk at a > higher bitrate than you send to the icecast server at. > > I am planning to add a feature that allows you to have n streams going > out, each for different bitrates (or alternatively different sample >

Hi All, I'm having a bit of trouble with my new *NOW setup. I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from SimpleSignal.com. Outbound calling works great, but I'm having some trouble with inbound calls. First, we would get the "the number you have dialed is not in service" error on inbound calls. After some googling, I found out that I needed

On Thu, 4 Apr 2024 14:28:16 +0100 Zaheer Abbas via samba <samba at lists.samba.org> wrote: > Hello everyone, > > Samba stopped authenticating AD users after minor upgrade. > > Environment: > - OS: CentOS 7 > - Samba Version: Upgraded from samba-4.10.16-15 to samba-4.10.16-25 > The problem is, Centos 7 will go EOL in about 3 months, at which point you will have to

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message "login incorrect". I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Hello everyone, Samba stopped authenticating AD users after minor upgrade. Environment: - OS: CentOS 7 - Samba Version: Upgraded from samba-4.10.16-15 to samba-4.10.16-25 Problem: Clients are unable to authenticate with Active Directory credentials, receiving a "password incorrect" error. Verification: sudo net ads testjoin shows a successful join. wbinfo --ping-dc confirms