similar to: Asterisk SIP security

i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '8888246463' rejected because extension not found. but the extensin existed -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail:

i create inbound confi my confi is: [incoming] exten=> 18888246463,,1,Dial(SIP/8003,60,rT) exten=> 6463,1,Dial(SIP/8003,60,rT) exten=> 18888246463,,n,Wait(5) exten=> 18888246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '8888246463' rejected

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk

Hi, Why are we getting message in the asterisk [Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601<sip:601 at 111.118.185.107>; tag=2f498fbd [Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601<sip:601 at 111.118.185.107>;tag=209a8aa9 Regards Deepak Bhatia --------------

Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

Not sure where to ask this, I've been googling and not finding much that is helpful. At work I've got a multi-threaded program targeted at Linux. It compiles on RHEL 2.1 and 3, and is targeted at 2.1, 3, and 4. UP until today, the binary built on 3 has worked fine on 4. But on RHEL4 update 4 it dies a horrible death in pthread_create. I have reproduced the problem on Centos 4.4, where I

hi, I saw that Mike Dransfield tried to port 'state' plugin from beryl. What does it do? It should be able to place windows, based on name, class etc., to specific viewports. I recently converted to compiz window manager from WindowMaker (used it for 8 years), and I miss automatic 'pinning' of specific windows to particular workspace (or viewport in this case). Mike's

I'm tyring to fit a relatively simple nls model to some data, but keep coming up against the same error (code follows): Oto=nls(Otolith ~ Linf*(1-exp(-k(AGE-to))), data = ages, start = list(Linf=1000, k=0.1, to=0.1), trace = TRUE) The error message I keep getting is "Error in eval(expr, envir, enclos) : could not find function "k"". I've used this

Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway

Deal All Asterisk Expert If this possible to Create Voice Recording System Beside Main Asterisk PBX?, so Call be handle by 1 Server and Recording by other server. 1. How to accomplish. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090325/d57465cd/attachment.htm

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Hi all, I'm under the gun to rid ourselves of the nt4 PDC which we currently use. options are A/D, samba, pgina. I really dig pgina, but dont think I can pass citrix credentials properly :( So I'm left with samba vs A/D. A/D is well,, ummmmm A/D, so tyring to avoid it. Currently have used Samba file servers for years. Have played with Samba PDC with2.2 Now, going to try

Just built myself a little test server with an Atom 230 processor in it and am quite impressed with it so-far. Wondering is anyones used one in anger for a VoIP platform? I'm after something with a bit more oomph than the VIAs I'm currently using that I can use in a small box (mini ATX size) without going full-blown Xeons, etc. Cheers, Gordon

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

Hello All, Im using OpenSSH 4.3p2 and tyring to scan a list of 40 machines in my network with ssh-keyscan utility. I used the following command, ssh-keyscan -t rsa -f hosts.txt The man page says that this utility displays the host keys irrespective of ssh or host is up/down and its working great. But in case if the scan stops at 30th host due to some protocol problems, the utility exits and

I almost afraid to ask - but, I have Samba installed and operational. When I try to login to http://localhost:901 I get a logon screen. I type root then the password. But it keeps comming back as bad usersnam or password. I know thjis is simple. But it's my fist time tyring to use samba and I cannot find any answers on the internet. Thanks for your help