similar to: Subscribe 403 forbidden

Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in the following line: Voicemail: 0/0 Shows no Voice mail but I did leave a voice mail at the extension. Any suggestion on what I should look for in my * setup. I am not worried about the 481 coming back for the other side yet. Once I get a handle on the Notify, I'll work on the 481.

Greetings, I am hoping that I am posting my question in the right place, otherwise please accept apologies. The problem that I am trying to resolve has to do with the network and is most likely a very trivial issue, but I simply cannot find any leads to solving it even after several carefully reading several threads on this mailing list and the Xen networking HOWTO. I am trying to setup XP as a

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

Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan

Patrick hi. Asterisk can do that, and you don't need VOIP lines. If you connect Asterisk to the net, and all employees have a VOIP phone (either hardware or software) then you're good to go. What do you need? To begin with, install linux on an old pc (well, not too old). Then go to voip-info.org and take a look at the Asterisk wiki. Everything you need is there. And of course, we're

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

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 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

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

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:

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

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

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'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