search for: dissallows

Hi there, I have tried very hard to setup the x-lite with asterisk, but until now i didn't get sucess. When i start the asterisk in debug mode, i see the message: sip/2.0 401 unauthorized. I know that this problem with authentication. I put in my sip.conf as below. [2203] type=friend username=2203 auth=md5 secret=1234 reinvite=no canreinvite=no dissallow=all allow=gsm

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"

Hi, sorry if it is the wrong approuch to suggest improvments to OpenSSH, but here comes my suggestion: I recently stumbled upon the scponly shell which in it's chroot:ed form is an ideal solution when you want to share some files with people you trust more or less. The problem is, if you use the scponlyc as shell, port forwarding is still allowed. This can of course be dissallowed in

Hi, I have my Sipura SPA-3000 setup with Asterisk as follows: [spa3k_line1] type=friend context=home secret=PASSWORD host=dynamic dtmfmode=rfc2833 dissallow=all allow=ulaw When an incoming call comes in, I have a Zap interface in Asterisk which just does a Wait,15 then answers with voicemail. The SPA-3000 detects the PSTN call and makes Line 1 ring - so I can answer the phone if

Hi, I am playing around with SIP extensions on my local lan using X-Lite but I am having a bit of difficulty, I have set up X-Lite and my sip.conf accordingly, but when I start it I get the following message: "Login failed! Contact Network Admin" I am still able to dial local extensions on my * with x-lite even though it is in this state, although trying to dial my sip extension from

I assign an email to the from attribute in my actionmailer, but I think because I use the TLS plugin to get actionmailer to work with gmail, it dissallows me to use change the from address. Instead it uses my gmail account. Has anyone experienced this before? -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails:...

I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me