similar to: I can't get access to the Compiz forums: 403 - Forbidden

For those who haven't seen it, the compiz forums are now at http://compiz.net/, along with the new spiffy logo. It'd be nice to have a basic web page/wiki there as well so people can upload documentation and so on. thanks -mike

Hi all, I have been worriyng and googling a lot but I can't find my mistake. I am trying to regiter an X-Lite Softphone to Asterisk, but I am getting a SIP/2.0 403 Forbidden response: SEND TIME: 10157385 SEND >> 10.100.249.12:5060 REGISTER sip:10.100.249.12 SIP/2.0 Via: SIP/2.0/UDP 10.100.249.86:5060;rport;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester

Download V 0.4 here http://sourceforge.net/project/showfiles.php?group_id=123387 burn it to an .iso install into asterisk box (be warned it deletes everything on the hard drive but this is what you want right :) it will automatically install Asterisk AMP FOP and Web Meetme read the FAQ here http://asteriskathome.sourceforge.net/faq.html basically if you are using a X100P all you need to do

I'm getting 403 forbidden errors when attempting to register to a certain provider. I've tried just about every combination of configuration settings I can think of with no luck. Following is what I would think should work (and one of the settings I have tried). Rather then list every combinaton I've tried, what are the common causes of a 403 forbidden on a register attempt? Other

Hi all, I have configured Asterisk using database. The real time is working pretty well. That is asterisk is picking up details of peers and the extensions as well properly from the MySQL database. -- Executing Wait("SIP/bharat-f720", "2") in new stack -- Executing NoOp("SIP/bharat-f720", ""Welcome to Asterisk"") in new stack

Hi guys, I'm using Centos 3.5 with Apache-2.0.46. i linke my mrtg from /var/www/mrtg to /var/www/html/mrtg so i did the command ln -s /var/www/mrtg. it worked fine last week but when i checked the mrtg today it say 403 forbidden. Forbidden You don't have permission to access /mrtg/ on this server. ------------------------------ Apache/2.0.46 (CentOS) Server but when i tried to link

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

I''m getting a 403 Forbidden error when I try to access the rails welcome page. I''ve followed the instructions for implementing rails on Apache2 at http://wiki.rubyonrails.org/rails/pages/Tutorial using an alias, but isn''t working. Can anyone help me out? Thanks. -Ofir -- Posted via http://www.ruby-forum.com/.

i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. ------------------------------------------------------ *CLI> odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI> ------------------------------------------------------ and user is added to

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Hi, I upgraded my local Asterisk (the last version was quite old), and since then, whenever anyone tries to call me via SIP/IAX thru my external Asterisk, they get "403 Forbidden" as soon as I pick up. I have no trouble picking up when someone calls via PSTN. Basically, my phone (Firefly softphone) will ring when they call, but will disconnect as soon as I pick up. It won't even

trying to set up and configure a polycom soundpoint ip 500 phone, when trying to get it to register with sip, i get the following message Sip read: REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58 From: "138polycom" <sip:5622832456@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To:

Hi, I have two softphones connected to an Asterisk "stable". I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a "403

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the

Hello, I have a question. I set up an extension to 1234 exten => 1234,1,Dial(SIP/1234@sip.greentone.com:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Morning all Am in the process of testing a migration of Puppet 3 from webrick to Puppet. Have found the foreman modules (https://github.com/theforeman) which seems to take care of a lot of the leg-work... However having got Puppet running with Passenger in Apache, whenever trying to access the Puppet master from a client, I was getting a ''403 Forbidden error''. Have dug

After several weeks of discussion the leaders of Compiz and Beryl have agreed that the two communities shall reunite. This decision is supported by both David and Quinn and represents the majority decision of the administrators and developers in each community. At this early stage not a lot has been decided, but these are the main points of the agreement: - As had previously been decided,

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden