similar to: Connection two asterisk via SIP (call forward)

I'm trying to get running an ethernet with two machines: a samba server running suse linux 6.4 and a client running windows 98. I follow the steps in the DIAGNOSIS.txt file, but I get stucked when executing 'net view \\SERVER' in the client. I have laready done everything it says in that file: 1) Set a lmhost file in the client. Just with this line: 192.168.0.1 ASTERIX #Asterix is

Greetings, Some users on a Windows XP Professional host are (lately) unable to use their roaming profile. A Windows error message states that due to a security problem or a corrupt profile, that it is unusable. A second error message immediately following the first states that a temporary profile will be used and that any changes will not be saved. Other users are able to log on with their

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

> Hi folks, > My SMB server is a Helios Ethershare 3.1 on a Tru64 Alpha machine. > We are now integrating more and more gentoo-Linux Servers. These are the > SMB Clients. > > On Obelix (one of our gentoo-Servers) I can mount the SMB Volume from the > Ethershare Server. > obelix nc4smb # whoami > root > obelix # mount -t smbfs -o username=m.heckmann

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

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

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

Hello, I've got a problem using the plugin managsieve in dovecot (1.0.10 on gentoo linux). I use local unix accounts on a linux machine, one for each user with a .maildir in there home directory. Here is a snippet of the logging. Feb 14 20:17:48 obelix dovecot: Dovecot v1.0.10 starting up Feb 14 20:18:00 obelix dovecot: imap-login: Login: user=<michael>, method=PLAIN,

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

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 all, A week or so down the list, i read that not many people were using realtime on an Asterisk18, so i had this afternoon a go at it... [sorry for the inconveneant line-wraps] First i did: mysql> create database asterisk; mysql> grant all on asterisk.* to 'voipadmin'@'localhost' identified by next i used the info from the wiki: CREATE TABLE `sip_devices` ( `id`

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 =>

[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

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

I seem to be having a problem with B3 on my ISDN line, as you can see from the dial string I am having to have asterisk generate ringing else there is no progress indication. -- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/ b|40|r") in new stack -- Called g1/142392203000/b -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/

Hi! I have some problems with the download of Asterisk-addons. I try to follow instructions that I found in http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but nothing to do. This is my shell: [root@obelix root]# cd /usr/src [root@obelix src]# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot [root@obelix src]# cvs login Logging in to

Is there a reason the variable ${DNIS} does not get set with incoming calls via chan_capi ? Is it related to the MSN=X in capi.conf ? version = chan_capi-cm-0.6.3 example; exten => _95555555XX,1,NoOp, ${EXTEN}, ${DNIS} == ISDN1: Incoming call '0400000000' -> '95555555' -- Executing SetCDRUserField("CAPI/ISDN1/955555 55-135", "Incoming") in new