search for: cayona

...ose the real issue would be the system you are planning on installing it in. I think that either of the cards are rather rough on a systems pci bus so faster is better. If you are just wanting to play and experiment I would go with the clone card and see how it works. Your milage may vary, Steve Cayona

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

...phone registers with the server ok but cannot make calls, or you'll have one-way voice issues, etc, etc. If you need some help getting it set up this way contact me off-list and I'll give you a hand. I've done it several times this way and its not really that hard. Regards, Steve Cayona Date: Sat, 18 Mar 2006 08:43:00 +0100 From: Anthony Azzopardi <tazzopardi@camline.net.mt> Subject: [Asterisk-Users] I have my asterisk machine behind a Linux Nat ... To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <441BBA0...

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a

Hi, I have a problem with asterisks on Linux. Looks like it is a iptables problem. My external client (eyebeam, on a different computer) cannot register to the asterisk server, but the asterisk server itself *looks* working. If I dial one of the incoming phone numbers for the server, I can see the call arriving in Asterisk (using asterisk -r). I tried nmap on my server, and this is the result:

I have a DID vendor that wants me to be able to generate specific SIP error messages under certain conditions and I'm completely stumped on how to do these: #1 - They want to see a SIP 503 error response(service unavailable) when they send the call in to an active extension and and the service is not available, I don't have a clue on how to simulate this. #2 - When they send in a