similar to: Problems with realtime sip

Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have

Why do attacks from the Internet get shown in the Asterisk logs with myAsteriskServerIP instead of the attacker's IP?! Really useful for blocking them, that is... Example: [Mar 6 00:00:00] NOTICE[1926] chan_sip.c: Failed to authenticate user 5550000<sip:5550000 at myAsteriskServerIP>;tag=ab8537ae (I replaced our IP address with myAsteriskServerIP. The attacks are not coming from

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

On 19/01/2015 4:13 PM, Nathan Kurz wrote: > On Mon, Jan 19, 2015 at 1:00 PM, Felipe Balbi <balbi at kernel.org> wrote: >> I just thought that such a small patch which causes no visible change to >> SVN users and allow for git users to build R would be acceptable, but if >> it isn't, that's fine too. > > Felipe --- > > It would appear that you are

Dear all, I have a question with regard to the use of hclust. I would like to be able to specify my own distance matrix instead of asking R to compute the distance matrix for me. It is computationally easier for me this way. My question is: How can I get hclust to accept this? Thanks, Ranjan -- *************************************************************************** Ranjan

Hello! I tried scanning through the 100's of messages in the last month.... I saw a post where you could call through the * server to FWD. Can someone help me out with this? Some direction. Thanks, Bill Flood

Hello. My problem is Rsync stops when I use it between 2 of my servers (2 NAS Synology) ( named "*.22*" and "*.6*" ). The problem continue... For example : _ Rsync run correctly between my server "*.22*" and ".6" ( in the 2 directions ) _ Rsync run correctly between my server ".6" and "*.8*" ( in the 2 directions ) _ Rsync *doesn't

Hello. My problem is Rsync stops when I use it between 2 of my servers (2 NAS Synology) ( named "*.22*" and "*.6*" ). For example : _ Rsync run correctly between my server "*.22*" and ".6" ( in the 2 directions ) _ Rsync run correctly between my server ".6" and "*.8*" ( in the 2 directions ) _ Rsync *doesn't run* correctly between my

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

This probably is a postfix problem, but I think there are lots of postfix experts/users on this list, and have heard a lot good things about this list, so I am just giving it a try. Thanks in advance! I am using dovecot-1.0.0-8_56.src.rpm downloaded from atrpms.net, and rebuilt from it(rpmrebuild ...). Postfix is 2.4.3. I followed documents at http://wiki.dovecot.org/LDA and LDA/Postfix, and

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 --------------

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

We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA <sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1 at

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

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

Hi, This is regarding Dtrace usability for memory leak detection. We have real-time application written C++ which runs on Solaris 10 having a problem that''s the my application grows in size from 130 Mb to 450Mb in around 15 days. So there is two possibilities with the application growth of memory due to Size growth of Dictionary Objects (Like Maps) and Memory Leak.

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 every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but

Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100<sip:100 at 108.161.145.18>;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you