search for: elmegs

Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures & the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no

Hello, I have Elmeg IP290 phone and have problems with VM. I don't know how to configure this IP phone, that it could call to *97@192.168.0.1 if I pressed "VMail" button. Now if I press buttom "VMail" , ip phone dials: sip:asterisk@192.168.0.1 (192.168.0.1 - Asterisk IP). So I don't understand , from where it takes "asterisk", cause I have never write

Hi! I have a little problem with the given phone... Do someone know it? My problem is that I'd like to display the name of the caller (if it is saved in the address book, of course), but it always display just the number... Thanks Luca Bertoncello (lucabert at lucabert.de)

Wondering if anyone on here can help with a niggling issue: One of our extensions is unable to make attended transfers at all. The phone in question is an Elmeg ip290, and works fine for direct transfers. However, on attempting to make an attended transfer, the first leg succeeds (the inbound call is placed on hold and gets MoH, the Elmeg user announces the call to the target extension), but upon

Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity

Hi list, I am in the process of setting up Asterisk for a new office and since this is going to be my first "real" installation I'd appreciate some advice on the hardware from the real world. We will have 8 channels (still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely go for a Digium card with echo hw cancelation) and a DSL 2mbit line (QoS on the switch and

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone.

Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the

> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 22:56 schrieb Antony Stone: > > Hi again, > >> 2b. Take your Thomson telephone to some other location with Internet access, >> let it register to your home Asterisk server, and them make a call to the same >> number yet again. I'm sure you can get

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello: Hi So, I got a phone (Elmeg IP290) from a collegue and tested it... > What I'll do tomorrow with a test phone is: > > 1) connecting it to my Asterisk and try to make a call > 2) connecting it directly to the servers of Deutsche Telekom (using my > network) and try to make a call Absolutly *no changes* on the behaviour

Hello, We found out that after upgrading the firmware in our GrandStream BudgeTone phones, that we were not able to transfer calls anymore. We use the BT's own tranfering mechanisme. We can dial the phone where the call should be tranfered to. But after that, the original caller stays in music on hold on the server and there's no way to get the calling channel back again (not to the

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to test there quality and feature set. So far we have a Grandstream 2000 Grandstream HandyTone 488 Cisco 7912 Polycom SoundPoint IP And we are looking at getting a Linksys SPA-942 Anyone have a favorite? -------------- next part --------------

Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal

BUG/SYMPTOMS: 1.Under certain circumstances, octoBRI (and most likely quadBRI) ISDN cards (Junghanns/CologneChip) severely distort certain ISDN payload. 2.Although these claims relate to the bristuff patch, the problem might not be limited to bristuff and in fact be rather asterisk/zaptel/libpri related. 3.Signal distortion is limited to the use of the CVS version of the Bristuff patch for

Hi, We are working on a product to analyze pcap files of VoIP calls. So far it does a reasonable job of analyzing the frequency distribution of packets in both directions, pointing out which direction packet loss / bad jitter occurs.  If you can trap the traffic on the outside and the inside of your Banana Pi and send me the pcap files, I would be happy to run it through our analyzer as

Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony > You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to

Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently