similar to: Strange SIP Activity

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

Good morning, I've been having some problems getting the SPA-3102 working properly in India. Specific problem is that calls from the Asterisk server out the FXS port is failing. When trying to make calls, I'm getting this message: [Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call from '' to extension '66200' rejected because extension not found.

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

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

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

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

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

On 2018-09-09 10:27, Antony Stone wrote: <snip > 1. Try removing one of the two commas. > > 2. Take a copy of your dialplan, and then strip out *everything* except > the > one context and the one number you want to match: > > [0705680837] > exten => 31705680837,1,NooP( Incoming 31705680837 on CC) > same => n,Answer(); > same =>

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited)

-------- Original Message -------- Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound Date: Sun, 06 Mar 2005 19:11:22 -0500 From: MF Hulber <mark@hulber.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, dan@mirrorlynx.com References: <200503060703.XAA12457@comand.net>

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

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You will need to add the following line before running > asterisk. > > ulimit -n 32768 > >

Hi I have a problems with a new Asterisk Server, when i want call, i have: [Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160 handle_request_invite: Call from 'PHISIP000001' to extension '00420225352184' rejected because extension not found. but into my extensions.conf: exten => _00420X.,1,Set(CDR(CodeTier)=CZE) exten =>

Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. Thanks in advance, -Motty -------------- next part -------------- An HTML attachment was

I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension

On Monday 10 September 2018 at 21:54:33, Marcelo Terres wrote: > I have think it should be > > context=0705680837 > > Not > > context=[0705680837] Ha! You're right... so simple :) Antony. > On Mon, 10 Sep 2018, 20:43 , <asterisk at a-domani.nl> wrote: > > On 2018-09-09 10:27, Antony Stone wrote: > > > > <snip > > > > >

Hello I am currently having a problem, that threatens to drive me insane... I cannot understand how Asterisk matches up a sip request with a peer. Here is my example: INVITE sip:87654321@192.168.100.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.59:5060;branch=z9hG4bK00d97b99;rport From: "1088200336" <sip:1088200336@192.168.100.59>;tag=as54af3e4d To: <sip:87654321@192.168.100.4>

Hi all, I think i understand the problem, actually I have two asterisk server. In the extension.conf file on one server I have added exten => 3923903,1,GOTO(s,1,3923903.conf) which reads the corresponding conf file when ever the extension no. through PSTN is called and learns the location of inbound.php which contains the IVR script to be executed. Now what i want is that through this

I've got PBXNSIP running on a windows box and it is trying to register with my Asterisk box. I can set up one trunk and it works fine, however if I try to setup a second trunk from the same box, there is some sort of authentication issue where Asterisk appears to be confusing which trunk is which. Here is the chunk from my sip.conf: [TEST1] context=STUFF-LD type=friend