similar to: FW: Under heavy attack

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

Hello All: Using the below dataset how can I make a barplot with Date(X) and NumEggs(Y) by Site. Then plot Temp(lineplot) It seems really simple, but I am having a hard time trying to do it by Site. Thanks Date NumEggs Site Temp 1 2008-04-22 0 Massacre Flat (RK424.5) 51.20 2 2008-04-23 0 Massacre Flat (RK424.5) 50.80 3 2008-04-24

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I'm trying to install both a Sangnoma A102 (with echo cancellation) card and a Digium 8 port analog card with echo cancellation (Digium AEX800E) in the same server. I know I probably shouldn't have mixed vendors - lesson learned for next time. That said, I have everything working fine...except Native Bridging between the Sangnoma and Digium cards. When I do native bridging, I get a very

I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI

We need to monitorate the E1 with nagios, somebody did this? any ideia? Thanks in advance! -- Atenciosamente, ------------------------------------------------------- Dario Quiroz (71) 9275-9080 gtalk: darioquiroz at gmail.com ------------------------------------------------------- -------------- next part -------------- An HTML attachment was

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101029/f6bd1f95/attachment.htm

Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)? As well, when I "SIP SHOW CHANNELS" I see phones registering showing as channels in use. Is there a way to filter this output? Thanks! MD

Hi, I am facing issue while generating a dial plan for the following case: all caller should be asked a code to enter than All the callers should be connected one extension. also tell me testing scenario : I have pbx setup and currently I have soft phones to use as extension. Currently I have created a dial plan using vdp I tried submitting it here but I don't know how to extract text

Hi I'm sorry for the my trivial quest. I Have asterisk 1.4 with TDM 400 with FXO and FXS, and works fine from several months. Now I want to connect a device to TDMFXS that want a ring frequecy of 25 hz to activate: i am italian, and usually the ring freq is 20 hz. The other time (I have used that device several times with other asterisk installation) I have modified /etc//modprobe.conf and

Hi ? I ?wonder if?anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT? - NAT - Server Client can hear users from server side but server cant hear client. ? Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no ? Ive check the sip headers in the debug mode and its using the external address in

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring

Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm