similar to: 403 forbidden error

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

Hi all, I have situation in that we are renting out a room in our office building to another company, including the provision of phone lines via our Aterisk box. To keep billing simple, we would like them to be billed separately for all their calls. We think we can achieve this by them leasing their own outbound IAX trunk (which our providers have said is no problem) and having all their

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

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

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/.

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

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home. We have also added new version of the usual Asterisk software AMP

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

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 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

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:

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 list I'm trying to get access to the Compiz forums, but I am greeted with a "403 - Forbidden"-page. Is this a known issue? Regards, Rune

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

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

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

I am using nginx with unicorn as a reverse proxy. Has been fantastic, but needed to look through some code for first time. Using ruby 1.9.3p194 with rails 3.2.9, development mode with ssl enabled, I start unicorn with -d ...and it does not get served up by nginx. Any ideas, or more information needed? Thanks! Jet Below is my unicorn.rb for developent: # config/unicorn.rb env =

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 ;-----------------------------------------