similar to: Conf files doubt

Hi guys, I create a topology like fellow: /****** /************ /*********** * GK *-----------* Asterisk *---------- Sip Prov * ******/ ************/ ***********/ | | | | | | H.323

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

hi, just change the problem in another email. I can make calls from H323 to Sip, but cannot on the other hand, i.e. from SIP to H323. I am thninking that the asterisk don't response the REQUEST of the Sip Provider. Or, the GNUGK don't do a response to the asterisk that don't malke a response to the Sip Provider!!!! AnyIdea!?!? Thanks in advance. Giscard

Hi there, I know; maybe you did answer to this question before but i am new in this mailing list... I have 3 GNU/Linux (Slackware) installed machines and 2 Windows 2000 installed machines. One Slackware machine is a file server... I have create 7 users in this server and i want to set my samba configuration file for share thems home directory and some group directories... When it works fine,

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

I wonder if anyone can help me? Am trying to terminate to H323 Vegastream. I'm using OH323 with little success. I can dial out and answer but voip end just keepings ringing and ringing. Thanks for any help. Neil Config file: [general] listenAddress=ALL listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no jitterMax=100

I'm writing to ask if R Core would make file.exists more Windows tolerant when the argument has a trailing slash. This has been discussed by users a few times here, I know it is not a new topic. But it is not a solved problem, yet. I acknowledge that CRAN packages exist which fix this by replacing file.exists(), but it seems more elegant to me to fix the problem in R itself. R Core goes to

Hi Timo, I have another problem, this time with user shared folders: User "markus" shared the folder "ForTest" to test: SETACL "INBOX/ForTest" test akxeilrwts dovecot-acl and shared-mailboxes.db have been successfully updated. As user "test" the folder (and "#User" namespace) is not visible. When I configure the shared namespace with

When sending calls to my Long Distance Provider I've come across this problem. After about 3 or 4 seconds into a call, it gets cut off. This is what I have concluded after doing a trace. 1. An invite is sent to the Asterix PBX 2. Asterix sends back a 100 trying. 3. Asterix then sends a 200 OK, with session description. 4. They ACKnowledge the Asterix 200 OK 5. Asterix then sends a 183

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7

i have a lot of questions, maybe you've got some answers. I've got the following setup : 1 central server ( linux ) , 17 servers ( LTSP ) on different locations, where each server has a number of thinclients connected to it ( max 6 ) The central server / 17 LTSP servers are on the internet and share 1 VPN ( freeswan ). It would be nice if the 17 LTSP servers could use asterix on one

Hmm, I don't have /etc/modprobe.conf, and wctdm is giving me problems. Which device is it talking about? [root@asterix /]# modprobe wctdm /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Hi: Just observing that there's been no trumpet blowing about Icecast 2.1RC1 being released, with no link on the website or nothing. For thems that want it, go to http://svn.xiph.org/releases/icecast Also, I presume that this release has the fallbackoverride functionality in it. Probably good to have this in the conf file. This release comes just at the right time for me - I will be

Hi Guys, I know no doubt this has been covered on the list a zillion time before, but can anyone point me to some good resources on using Asterix as a VoIP gateway? I would like to get two TDM400P cards in two machines attached to separate adsl connections (in two different physical locations). I'd then like to be able to plug POTS telephones into each of them (I understand I will need FXS

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

New to Asterix and perhaps someone can help. The plnned configuration is that the Quintums are to register to the Asterix and the signalling to be handled by the Asterix but the media (G 729 code) to be directed to the service provider. Thanks Shaun

Hello, I inherited an Asterix phone system. I am well versed in Windows based platforms but have zero experience in Linux and Asterix, no make matters worse I have no documentation on this system. I had to change the entire networks gateway address for various reasons but now the Asterix system will not send messages via email. I think it is because of the gateway change. How do I change the

Hello. I have a small problem when I want to display a dendrogram for about 450 varieties. The label of theses varieties is too big and we can not see thems well. How can I do to resize the label smaller? Can you help me please? Thanks. [[alternative HTML version deleted]]

Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone