similar to: Asterisk not logging originating IP of a brute force attack

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

Hello, I have been trying to get a TrippLite Internet Office 750 UPS to talk to my Linux PBX for a couple of evenings now and I'm getting nowhere... I tried searching the list archives before posting here (I'm sure I'm not the first one to try to get this going) but they seem to be offline... The UPS is unfortunately USB based but I thought I'd give it a try anyway. Here is

Hi All, I am using asterisk 1.4 branch on server. Here is a my dialplan. i have set the incoming route to incoming context, and then i have set dial with local channel, The call comes to my server and the call is routed to matched case, so my phone 1001 ring for 30 seconds. If i got the NOANSWER then the channel is not passing to next priority. I need to pass that channel to the next priority of

Ronald Wiplinger wrote: > Steve Totaro wrote: >> Ronald Wiplinger wrote: >>> kevin ling wrote: >>>> Hi, >>>> >>>> Check the vm_general.inc file >>>> >>>> >>> Where should this file be? >>> >>> >>> bye >>> >>> Ronald Wiplinger >>> >>> >> You

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Hello Using Asterisk 12.8.2. I now have the "via ICE" messages in the RTP debug (see below). If you look in the SIP debug (see below), you also now see the "ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the webRTC client. But still no audio ! None at all ! In both directions. You can see in the SIP debug that the IP-address in de

Hi there my zaptel hardware is giving errors while loading but they seem to load just fine. the lights wil work and my wctdm card is also workin and the isdn works to But when I stop asterisk I have to reload al cards again is this normal? This is my zaptel.conf is there no way to group these because my te110p is giving an error that it can't find channel 35 but 35 belongs to my wctdm.

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [root at mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [root at dreampbx ~]# ps aux | grep asterisk root

Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site:

Could anyone outline a method for trouble-shooting these messages "SIP/2.0 482 Loop Detected" I'm seeing on a particular peer? There is no call going on when these pop up. Sip read: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 207.149.XX.XX:5060;branch=z9hG4bK0a322471 From: "asterisk" <sip:asterisk@207.149.241.3>;tag=as395506d6 To:

I use smbmount to mount a number of NT shares from around a global network in order to sync from a master distribution which resides in linux. Specifically... uklnxsrv$ smbmount //NTSRV/share -c 'mount /tmp/temporary_mount -u 502 -g 510' -U guest%guest_passwd Frequently it works well, but from time to time the mounted NT share, for lack of a better word gets corrupted. Here are the

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is an automated response to e-mail sent to security-alert@hp.com. All potential security issues submitted will be carefully reviewed by the HP Software Security Response Team (SSRT). A follow-up message will be returned to confirm that we are investigating and if necessary to request additional information. Any non-security issue or

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is an automated response to e-mail sent to security-alert@hp.com. All potential security issues submitted will be carefully reviewed by the HP Software Security Response Team (SSRT). A follow-up message will be returned to confirm that we are investigating and if necessary to request additional information. Any non-security issue or

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is an automated response to e-mail sent to security-alert@hp.com. All potential security issues submitted will be carefully reviewed by the HP Software Security Response Team (SSRT). A follow-up message will be returned to confirm that we are investigating and if necessary to request additional information. Any non-security issue or

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

Hello ! I have two lans (LAN A & LAN B) connected via freeswan vpn gateways (VPNGW-A & VPNGW-B) over DSL. Logged in at VPNGW-A, I can only connect to the clients *behind* the VPNGW-B but not the VPNGW-B itself. ---- example VPNGW-A# ssh -l remoteuser 172.20.7.1 [IP 80.133.197.130.40523 > 172.20.7.1.ssh: SWE 343226483:343226483(0) win 32440 <mss 16220>] ---- no connection I