similar to: Incoming SIP calls with no extension

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

Hi, Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? I've got an server with an Asterisk and the following cards: 1. TE110 -- to telco 2. TE400P with one FXS to analog phone 3. Two HFC-S based cards in NT mode I'd like to connect ISDN modem to one HFC-S card and allow dialout through TE100 to some external number (being more precisely, to some

Hello, I just bought a Snom-320 from ATAComm. I plugged it into my LAN, registered it with *, etc. All my other SIP gear is Sipura and works fine, both on the LAN and over the Internet. The new Snom seems like it can't process the audio from the handset mic. A steady tone is garbled, even on the LAN. I've contacted ATAComm, Snom and the company representing Snom in the US. So

I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from

Hello, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. * is registered to the innovaphone gatekeeper. Trunk connection is done with an AVM-B1 and chan_capi.

Hi There, I have an asterisk an a Digium 4 Port E1 Card On E1 Port No. 1 I have the Telekom PRI On E1 Port No. 2 I have an Alcatel PBX that cannot be changed So I have setup my asterisk between Alcatel and Telekom In extension.conf i configured: [telekom] exten => _9149.,1,Dial,ZAP/g2/${EXTEN}; exten => _9149.,2,Hangup This works great, all incoming calls are directly routed to alcatel

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call

Hello List, is someone operating a DID /P2P / Anlagenanschluss with more than one HFC-Based ISDN-Card ??? I have now 12 hours of setup-troubles behind me with Colt-Telekom, where we did not get it working with two HFC-based cards. Here the setup: - 2 HFC-ISDN-Cards (the one from Conrad-Electronic) - bri-stuff.0.0.2 (with the asterisk-sources from the download.sh-skript) - two NTBAs from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

Dear List, I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an "Anlagenanschlu?" with 8 B-channels in Germany. It worked fine with Deutsche Telekom, but since we switched to Arcor nothing works at all. After some debugging, I called Arcor helpdesk who told me that they do not have a D-Channel and cannot signal one at my request either. Is it possible to use

Hi list! I'm very new in Asterisk and VoIP, and of course I have a problem... :) Well, my problem is, that Deutsche Telekom wants me to change my ISDN to VoIP... :( I must do that, since I have no alternative. Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can configure my two numbers by Deutsche Telekom and I got now an extra number from Messagenet.it. Now the

Hi list! I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didn't changed my Asterisk configuration). The problem: after 15 minutes will the call dropped, but only if the call is to another nation! If I just call another phone in Germany, I can speak longer than 15

Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new numbers. But i always get: Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel 'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no invalid handler I only changed the

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, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a "0" as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success:

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with

Hi list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've done that to improve security and to be able to assign all calls coming in

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.