similar to: Asterisk 1.8.22.0 built - encrypt authentication

> 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 =>

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:

Hi all, I'm having issues with overwrite caller id, when I call someone my caller id should be "mycompanyinc" but instead my id shows up as my extension number 101. this is what i have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid="iuser 101" disallow=all allow=ulaw allow=alaw username=101 secret=Passwd dtmfmode=rfc2833 host=dynamic mailbox=101 at

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

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz <motty.cruz at gmail.com> wrote: > here is what I did and worked for me: > > exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) > > exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have a leading underscore on your patterns. Your users

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 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

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, 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 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, 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>

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

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.cruz at gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz >>> I continue to see errors like this: >>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: