search for: elmeg

Displaying 18 results from an estimated 18 matches for "elmeg".

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2005 Mar 04
4
Difference between Snom 190 & Elmeg 290?
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...
2007 Jan 31
0
ELMEG IP290 and voicemail
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 un...
2020 Aug 30
1
[OT?] Elmeg IP290: do someone know this telephone?
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)
2007 Jan 15
0
SIP transfer issue
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 completing the transfer, both parties get MoH, not each othe...
2007 Feb 07
3
Diagnosing poor call quality
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 co...
2006 Apr 14
2
Asterisk hardware for new office suggestion
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
2020 Jun 14
4
Voice "broken" during calls
...l other network services on my Linux-Box Firewall/Gateway, including the traffic shaper (in the case, this was the problem), then connect my Thomson phone to the Telekom's server and call my father in law. Always the same problem... So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290). I'll connect it to my network and my Asterisk and will try to call my father in law for a test. I really do *not* expect any change in the situation... I think, the problem should be somewhere by Deutsche Telekom... What is your opinion? Btw: I did all tests with my father in law, s...
2020 Jun 14
3
Voice "broken" during calls
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
2020 Jun 14
0
Voice "broken" during calls
...> Linux-Box Firewall/Gateway, including the traffic shaper (in the case, > this was the problem), then connect my Thomson phone to the Telekom's > server and call my father in law. > Always the same problem... > > So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290). > I'll connect it to my network and my Asterisk and will try to call my > father in law for a test. > > I really do *not* expect any change in the situation... I think, the > problem should be somewhere by Deutsche Telekom... > > What is your opinion? > >...
2020 Jun 15
0
Voice "broken" during calls
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...
2005 Jun 01
0
BT101 new firmware problem (1.0.6.3)
...he server and there's no way to get the calling channel back again (not to the first receiver, nor to the transfering target). At first I was thinking it had something to do with the asterisk, because upgraded to version 1.0.7 a week ago. Though, the strange thing is that we also have some elmeg/snom190's and they do not have this transfer problem. Not being able to transfer calls is a major problem. I'm puzzled. I didn't thought upgrading a firware would distroy existing functionality. The new firware version in the BT's is 1.0.6.3, the old firmware was still one of t...
2007 May 16
6
SIP Hardware Phone
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
2006 Jun 28
6
Suggested Phone
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 --------------
2008 Dec 11
4
Asterisk dies when external access is lost
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
2005 Aug 23
0
[Asterisk-Dev] Severe ISDN signal distortion in CVS-HEAD with octoBRI
...SETUP: The following hardware and software has been used to test the problem: - 1.7GHz Intel P4 / 512M - CologneChip octoBRI ISDN card (not tested with Junghanns.NET versions yet) - Debian Sarge / 2.6.12 kernel - Asterisk (CVS-05.29.05 / 1.0.7 / 1.0.9) + Bristuff (RC8f-CVS / RC8f / RC8n) - Elmeg and DeTeWe OpenCom PBX as analog channel banks. TEST RESULTS: asterisk + bristuff version: ISDN L1 Bit Error Rate: 05-29-05 + RC8f-CVS >1% 1.0.7 + RC8f 0 1.0.9 + RC8n 0 SITUATION/BACKGROUND: I'm new to asterisk devel...
2020 Jun 15
4
Voice "broken" during calls
...190* <https://www.facebook.com/jeff.lacoursiere> <https://linkedin.com/in/jeff-lacoursiere-884361> <https://www.twitter.com/stratustalk> On 6/15/20 11:55 AM, Luca Bertoncello wrote: > 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 &gt...
2020 Jun 13
3
Voice "broken" during calls
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
2005 Jul 04
3
Call Transfer using SIP clients
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