similar to: chan_capi question

Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients we

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mohammed Firdosh Nasim Sent: Tuesday, March 15, 2005 11:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * On Sat, 2005-03-12 at 07:42, Luki wrote: >

Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients

Hi All, I have two SIP softphones(Windows Messenger) running on different subnet (Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security issues both the subnets are completely isolated ( U cant even PING from one to other) and I want to connect Phone-1 & Phone-2 to the *. How can I proceed? Please

I just downloaded chan_capi.0.3.4a.tar.gz but it will not compile on my system (Suse 9.1). I compiled and installed Asterisk and it is running, did I miss a configuration or dependency for Chan_Capi somewhere ? cheers, Mike linux:/usr/src/chan_capi-0.3.4a # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES

Hi everyone, Hi, I am new to the ISDN world, today I received my new NT box from CS Telecom but unfortunately it doesn't work. I have installed all the drivers and when I issue a "capiinfo" I get: -------------------------------------------- Number of Controllers: 1 Controller 1: Manufacturer: AVM GmbH CAPI Version: 2.0 Manufacturer Version: 3.101-02(49.18) Serial Number:

Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the latest chan_capi (also tried an older version). When using the Capi-Channel, everything works fine except from the sound it sounds extremely choppy and is unusable :-( When e.g. capisuite is used for fax, everything

Dear asterisk users! I want to control a hardware pbx with asterisk. What I need to do this is being able to press "hold" which can be done with capicommand(hold) and then send digits on a bri card which connects to my asterisk computer. So far I use Dial(CAPI/ISDN1/27:<<digits>>/bo,15) to do this. Are there better ways? Note that these are not dtmf, I'm afraid. I use

Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple

Hi, I am having a problem with (Asterisk is crashing) with a Fritz card PCI / chan_capi. Receiving Calls from PSTN works, but outbound calls make asterisk crash (Speicherzugriffsfehler/Segmentation fault). The crash occurs upon dialing with the other phone not even ringing. I hereby ask if somebody reading this list can confirm or disprove my issue. Does anbody run a recent asterisk 2.6 with

Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr

Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is

I have a problem receving calls via the ISDN line, using the followin components Asterisk 1.0.9 with asterisk@home chan_capi-cm-0.6 AVM Fritz card datalink protocol = point to multimode I can make calls out with no problems so the issue is only incoming calls. When I make the call from an external line to the ISDN line connected to asterisk, I get a busy signal after about 5 seconds. I have

1. If the other end picks up the phone the conversation is successful and when the conversation ends, the channel is properly released: -- CAPI/ISDN1#02/2104988888-0 answered SIP/1000-081d91b0 == ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 2 == Spawn extension (internal, 2104988888, 1) exited non-zero on 'SIP/1000-081d91b0' > ISDN1#02: CAPI INFO 0x3490: Normal call

Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten =>

Hi All, Many thanks to Marc who helped me with a previous Capi Dialout plan - however..... What I now would like to be able to do is: - We have 8 msn's 383590, 383591 383592 etc. What I would like to do is set up an If Then Else type statement along the following lines: - If extension 7957 Then Dialout on Capi msn 383590 ElseIf extension 7958 Then Dialout on Capi msn 383591 ElseIf

Hi, I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens PABX. From a SIP phone, I can call other internal SIP phones, external numbers (to PSTN), but I can't call internal phones connected to the internal phone network. When I call 107, which is an internal phone, heres the logs from asterisk: -- Executing Dial("SIP/Greg-081f5a10",

Hi! I am playing around with Asterisk and have a problem :-) (Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a sip-phone at my desk and an ISDN-phone (independent of the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't

Hello everyone, I'm using very simple setup to make and receive external ISDN calls through a softphone (x-lite version 3.0 - Win32) via an asterisk box. Hardware setup: - Dialogic Diva BRI (lspci yields: Network controller: Eicon Technology Corporation DIVA Server BRI-2M/-2F (rev 01)) - ISDN BRI line Software setup: - Redhat 9 - asterisk-1.4.16.2 - chan_capi-1.0.2 Asterisk configuration

[Something seems to have went wrong with my previous posting. It appears on the archive page in another thread. I did not receive anything myself. So I may give it another try:] Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over