similar to: Asterisk 1.8.22

Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941" <sip:4941 at public_ip>' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '"4941"

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten =>

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

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, I continued to see this errors in the logs: [2015-12-02 10:05:57] NOTICE[19949]: chan_sip.c:23277 handle_request_invite: Failed to authenticate device 100<sip:100 at xx.xx.xx.xx>;tag=10cdeaf7 how do I guard against this kinds of attacks? Also, to get the IP address from where this attack come from I use the following command "tcpdump -lni eth0 -f "udp port 5060"

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>

Hello, I used this guide, it worked for me: http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html Thanks, On 07/06/2015 04:54 PM, John Kiniston wrote: > The Authenticate application will do this for you. > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate > > You can either give it a single PIN to use for all calls, Authenticate

Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: