similar to: ELMEG IP290 and voicemail

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

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

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

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, 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

Hello, My question has to do with a form resubmit problem. Say for example a form is submitted and within that form, a picture is uploaded. If the user presses the Back buttom and gets back to the form submission sheet, and then clicks on resubmit, once again, the same data is populated into my tables wasting space (another picture upload for no reason in a separate row). Now, I know there are

I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables

Dear helpers, In an attempt to use a loop to generate graphs in a for loop in run into a problem. The plan is to fill each page with eight graphs (mfrow = c(4,2)) in to two columns. Only the buttom graphs ( meaning every fourth graph) have tick labels on the x axis to preserve space. I used an if .... Else statement to achieve that. The problem is that the first eight graphs are skipped

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

Hi folks... I know, that perhaps this is not the right list for this question, but I need some help... I had install UBuntu with KVM. I deploy some Vm's with virt-manager. Everything works well, but when I use the control panel fro virt-manager, to force shutdown off a machine, some other machine that I do not select, down too... I click with right buttom from my mouse, select the proper

Dear helpers, In an attempt to use a loop to generate graphs in a for loop in run into a problem. The plan is to fill each page with eight graphs (mfrow = c(4,2)) in to two columns. Only the buttom graphs ( meaning every fourth graph) have tick labels on the x axis to preserve space. I used an if .... Else statement to achieve that. The problem is that the first eight graphs are skipped

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

> 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

On my 7905s I can configure a "voicemail" number, which in turn activates a "Messages" softkey. When pressed, it goes to that exten, which in turn is configured to go to VoiceMailMain. Great, works like a charm. However, the phones also have a "go to voicemail timeout" after which, the phone diverts the call to voicemail via a "temporarily moved". This

Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 -> voicemail -> Secretary 1 555-1235 -> voicemail -> Secretary 2 Any help would be greatly

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

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.

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

Hi, I can't figure out a way of achieving what I want to do with the voicemail feature. I thought I'd ask here to see if there are any creative solutions that I have not considered. What I want to do is have a message that says "Press 1 for Dick, or 2 for Jane". Then, depending on which number is pressed, have the caller sent to the appropriate voicemail box. I know how to