similar to: Asterisk and Eicom BRI-2M or 4BRI-8M

Hello all, I wonder to have some feedback on using ISDN BRI Cards with Asterisk and the Echo problem. I have tried a simple BRI card with i4l driver and encounter huge echo problem. I have tried to solve it with a Sw chocanceller without success. What I'd like to know is wether some of you have used other BRI Cards (I have seen reference to Eicon cards on this list) and if the echo

Hello, I would like to have some comparison between E1 cards from Digium and those from Eicon for a VOIP - ISDN Gateway. How does they compare on the echo cancel point of view? Is the echocancellation code for E400 good enough for production environment? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz -

Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE

Hello, Sorry for posting again my question about MGCP Phone and Asterisk But I can't use it. I'd like to know weather it is a pb of my confiuration (mgcp.conf), My IP Phone device or asterisk. I include my mgcp.conf file and may send some debug trace. Thank you for any feedback. Best regards, Daniel ANDRE ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr =

Hello, I have just received an MGCP Phone for test purpose and I can't place a call from my MGCP Phone. I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr = 0.0.0.0 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line => aaln/2 ;line => aaln/1 [192.168.10.10] host = 192.168.10.10 context =

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

Hello, Now that I have a nearly working configuration for my IP10S with * I wonder if anyone has done call transfert with this Phone. In the IP10S documentation they talk about the 'service key' wich is the key with the white dot on it. With this Key, it should be possible to have a menu with call transfert entries. This menu should (accordingly to the documentation) depend on the

Hello, I have just received an MGCP Phone for test purpose and I can't place a call from my MGCP Phone. I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr = 0.0.0.0 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line => aaln/2 ;line => aaln/1 [192.168.10.10] host = 192.168.10.10 context =

Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr) IRIS

Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr)

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen. I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine. So I am

Hello, I have an issue getting the chan_capi module to load in asterisk cvs from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva Server Bri card. I load the modules with: modprobe -v divas divacapi I load the firmware with: divactrl load -c 1 -f ETSI -vd6 Output in /var/log/messages is: Nov 9 19:26:26 voice kernel: Eicon DIVA - DIDD table (http://www.melware.net) Nov 9

for this process is "root", that is correct. The PID 7247 is one of the users, but the UID shouldn't be "root" and should be the username that was authenticated during logon. Anyone have same or similar problem? and how do I prevent this problem from happening again? I would appreciate any help. Thanks. Will Sun wsun@jpl.nasa.gov (818) 354-2311 Return-Path:

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: "call failed: declined". Anyone have experience with this

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

Hi (Armin?) ! Today I had a problem with Diva Server 4BRI-8M 2.0. Asterisk 1.2.12.1 chan_capi-cm-0.6.5 divas4linux-melware-3.0.f-106.622-1 Asterisk could not receive and make calls on the BRI ports, although the ports looked fine within Asterisk. I usually use "/usr/lib/divas/divactrl dchannel -c 1" to test line activity. This time there was no activity (cryptic log