similar to: Failed to authenticate device 100

Hi all, I continue to see the following msg on my Asterisk log: [Sep 8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite: Failed to authenticate device 9009<sip:9009 at 196.107.xx.xx>;tag=8dd48dd2 IP: 196.107.xx.xx is my asterisk server IP address. I don't know what it means and how to cover any holes that attacker is trying to exploit. Thanks, Motty -------------- next

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

Dear R-people: I tried to create a shared library in Windows XP. However I got error messages which attached below: C:\lasso>Rcmd SHLIB all.f cox.f gcc all.o libR makeMakedeps all.dll -o all gcc.exe: libR: No such file or directory gcc.exe: makeMakedeps: No such file or directory make: *** [all] Error 1 I have created shard libraries successfully before. Also for the same fortran files:

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

On 10/28/2015 06:37 PM, Pete Mundy wrote: > Hi Motty, > > Isn't the whole point of the nonce in a SIP registration to ensure the > secret doesn't go on the wire in plain-text? Is this not enough, or > are you looking to hide the username too? > > (if so, fair 'nuf, just wondering why :) > > Pete > > Ps, if so then I think TLS is the missing part of

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130509/24815a66/attachment.htm>

Hello, I hope to be in the right list. my question is the following. I can''t seem to get multipathing configure on XCP 1.6. In XenCenter I was able to enable "Multipath" however, in the LUN under general tab I see " multipath 1 of 1 active", the the CLI I do the following command xe --m node session it list only one session. Can someone help me please? Thanks in

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

Hello, I am searching for a solution to encrypt authentication from Asterisk server to clients. Searching srtp seem to encrypt traffic, I just want client authentication with encryption. Can someone point to the right direction? has anybody used ZRTP? experience with ZRTP? Thanks, _motty

here is what I have: exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: > Motty > Yes > From your dial plan accept 9 + 7 digits

Hello, I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3 Thanks, .Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150729/68b59f7c/attachment.html>

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -------------- next part -------------- An HTML attachment was

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

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When I called companies with auto animate menus my system does not seem to detect menus on ther other side. For instance I called this number (407) 886-3338 when I input the ext. number of any option on the list I don't get a response however if I called the same number from my google account or my cell phone number it works fine meaning I can select any option or input ext number.

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > >

Hello, recently I have seen spike in attacks on my asterisk server, this is what I get on the LCD of my phone: 201 at 76.220.5.205 or calls from 1000 sip1000 at 76.2230.5.205, have any idea on how to stop this calls? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am looking for CDR reporting solution? Any suggestions? I am using Asterisk 13.13.1 I would like a report on number of calls per extension. Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170322/cb7cd66f/attachment.html>