Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating With Asterisk
Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating With Asterisk
There are many ways to do this, and very little information to go on. For instance, if you have Exchange 2007 and a lot of money, you can integrate it with Asterisk using Exchange UM (Unified Messaging). Microsoft charges extra for the premium CALs you need to actually do that. There are also some pitfalls (Microsoft uses SIP over TCP, in Asterisk that mode is experimental. Asterisk usually uses SIP over UDP). Since your Windows application already exists, you must already have a way to generate voice mails for non-Asterisk systems. It is entirely possible, and in fact quite likely, that you can leverage whatever mechanism you are using for that. The more information you give us about your existing Windows application and how it interfaces with phone systems to begin with, the better information you will get. Also don't forget version information. Which version of Windows, which version of Asterisk, and is there any other software involved? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shyamala Devi Sent: Monday, November 08, 2010 2:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integrating With Asterisk Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joel Maslak Sent: Sunday, November 07, 2010 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Big practical systems I believe this looks like a standard channel bank. Asterisk generates all audio. Ring and hook status are sent out of band. Dial tones are in-band. Ringback, busy, congestion are in-band audio. I would think a standard T1 card would be fine. That said, I would verify this with the LEC. === Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed bit? Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integrating With Asterisk
Hi Kevin, Ours is an MFC application hosted on a .Net 3.5 Framework and it uses a third party SDK (MFC) for Voice communication. We don't have the code for it and the SDK is no more supported. Our application runs on Windows XP, Vista, 2003/2008 server Windows7, both on 32 bit and 64 bit platform. Regards, Shyamala On Mon, Nov 8, 2010 at 6:17 PM, Kevin Keane subscript...@kkeane.com wrote: There are many ways to do this, and very little information to go on. For instance, if you have Exchange 2007 and a lot of money, you can integrate it with Asterisk using Exchange UM (Unified Messaging). Microsoft charges extra for the premium CALs you need to actually do that. There are also some pitfalls (Microsoft uses SIP over TCP, in Asterisk that mode is experimental. Asterisk usually uses SIP over UDP). Since your Windows application already exists, you must already have a way to generate voice mails for non-Asterisk systems. It is entirely possible, and in fact quite likely, that you can leverage whatever mechanism you are using for that. The more information you give us about your existing Windows application and how it interfaces with phone systems to begin with, the better information you will get. Also don’t forget version information. Which version of Windows, which version of Asterisk, and is there any other software involved? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shyamala Devi *Sent:* Monday, November 08, 2010 2:55 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Integrating With Asterisk Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Sun, Nov 07, 2010 at 10:26:59AM -0500, Jeff LaCoursiere wrote: asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. It could be the echo canceller, I had this kind of problem with OSLEC. I also thought the PRI provider was sending clipped audio. I switched to the VPM450 daughterboard and since audio has been crystal clear. What is your setup for echo cancelling? -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get the Uniqueid of Action Originate in the AMI
Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a lots, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Saturday, November 06, 2010 1:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8 On 5 Nov 2010, at 15:04, Danny Nicholas wrote: Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to Machine 3, I get lags/pauses on Background/Playback commands. I play files and groups of files that last from 1-45 seconds, so I can press keys and proceed, but I don't expect my end-users to know to do this. Any clues? Do I need to open a tracker issue on this one? Thanks Danny Nicholas -- There is an open bug on this - or something very like it - https://issues.asterisk.org/view.php?id=18110 The work around seems to be to set internal_timing = yes in asterisk.conf and noload = res_timing_dahdi.so ;noload = res_timing_pthread.so noload = res_timing_timerfd.so in modules.conf Which forces asterisk to use the (older less efficient/accurate) pthreads timer. The bug looks to be being worked on, so I'm optimistic it will be fixed soon. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Thanks Tim, This workaround has worked for all of my testing this morning. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI
El 08/11/10 13:12, Rodrigo Lang escribió: Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Hi, If you are using the originate action in asynchronous mode, you will receive the uniqueid of the originated call in the OriginateResponse event, not in the response of the action. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI
Set event on while login into AMI and set your own uniqueid using action ID for that call . Example : action: login Username: your_user Secret: your_secret Event: On action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr ActionId: yourID Hope that will help. On Mon, Nov 8, 2010 at 1:12 PM, Rodrigo Lang rodrigoferreiral...@gmail.comwrote: Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a lots, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Lang Sent: Monday, November 08, 2010 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Get the Uniqueid of Action Originate in the AMI Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued If you generate this from an AGI, you can query the output and get the uniqueid from that. If you are doing it via a Call file or some other method, you are probably out of luck. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI
The other thing you can do is put UserEvent() calls in your dialplan that can
have pretty much anything you want in them.
exten = s,5,UserEvent(DidQueue,${UNIQUEID} ${CHANNEL})
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 8, 2010, at 10:45 AM, Miguel Molina wrote:
El 08/11/10 13:12, Rodrigo Lang escribió:
Hi to all.
I'm begin a use the AMI and i have the need to get the uniqueid from the
call i have generate using the Action Originate. Anyone can help me?
When I generate these commands:
action: Originate
channel: SIP/101
application: Dial
data: SIP/100,120,Ttr
The only response I get when the call is answered, is this:
Response: Success
Message: Originate successfully queued
Hi,
If you are using the originate action in asynchronous mode, you will receive
the uniqueid of the originated call in the OriginateResponse event, not in
the response of the action.
Regards,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
--
_
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Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI
Thanks a lot to all for the responses. I begin to use the event OriginateResponse, it's what i need. Thanks again. Best regards, -- Rodrigo Lang Opening your mind - Just another Open Source sitehttp://openingyourmind.wordpress.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Mon, 8 Nov 2010, Daniel Tryba wrote: On Sun, Nov 07, 2010 at 10:26:59AM -0500, Jeff LaCoursiere wrote: asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. It could be the echo canceller, I had this kind of problem with OSLEC. I also thought the PRI provider was sending clipped audio. I switched to the VPM450 daughterboard and since audio has been crystal clear. What is your setup for echo cancelling? I inherited this board, and don't think it has the echo canceller daughterboard. Is there a way to query for it without taking the machine down? It is loading MG2 otherwise. Using dahdi_maint -s 1 I am tracking a lot of errors, anyway, so have the carrier taking a look this afternoon. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Mon, Nov 8, 2010 at 1:44 PM, Jeff LaCoursiere j...@sunfone.com wrote: I inherited this board, and don't think it has the echo canceller daughterboard. Is there a way to query for it without taking the machine down? It is loading MG2 otherwise. 'dmesg | grep VPM' should tell you if you have a hardware echo can installed and activating. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DNS SRV
Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 Multiple Parking Lots
Hello, Recently we have been using asterisk 1.6 and everything worked ok, all our productions server are 1.6. Recently we have upgraded one to 1.8 and multiple parking call are not working, which worked pretty ok on asterisk 1.6. Are there any major changes in asterisk 1.8 related to park call and multiple parking lots. Bellow is my config from asterisk 1.6 which worked, I`ve tried this in 1.8 and it’s not working. features.conf [parkinglot_A] parkpos = 2011-2020 findslot = next parkingtime = 60 context = parked [parkinglot_B] parkpos = 2021-2030 findslot = next parkingtime = 60 context = parked [general] parkext = 2000 parkpos = 2001-2110 parkingtime = 60 adsipark = yes findslot = next sip.conf [user_A] ... parkinglot=parkinglot_A [user_B] ... parkinglot=parkinglot_B extensions.conf include = parked exten = 2011,hint,park:2...@parked exten = 2011,1,Wait(1) exten = 2011,2,Set(CHANNEL(parkinglot)=parkinglot_A exten = 2011,3,ParkedCall(2011) exten = 2012,hint,park:2...@parked exten = 2012,1,Wait(1) exten = 2012,2,Set(CHANNEL(parkinglot)=parkinglot_A exten = 2012,3,ParkedCall(2012) exten = 2021,hint,park:2...@parked exten = 2021,1,Wait(1) exten = 2021,2,Set(CHANNEL(parkinglot)=parkinglot_B exten = 2021,3,ParkedCall(2021) exten = 2022,hint,park:2...@parked exten = 2022,1,Wait(1) exten = 2022,2,Set(CHANNEL(parkinglot)=parkinglot_B) exten = 2022,3,ParkedCall(2022) With this config I was able to also monitor every parking place and see on the phone of there is a call parked on that extension and pick that up from every phone in the office and every user when they press “Park Call” are parking the current call to their parking lot. This config worked on asterisk 1.6 and we cannot make it to work on asterisk 1.8. Any ideas guys ? Thanks for the answers, Bogdan Sarandan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Multiple Parking Lots
Hi, On 11/08/2010 08:59 PM, Bogdan Sarandan wrote: Hello, Recently we have been using asterisk 1.6 and everything worked ok, all our productions server are 1.6. Recently we have upgraded one to 1.8 and multiple parking call are not working, which worked pretty ok on asterisk 1.6. Are there any major changes in asterisk 1.8 related to park call and multiple parking lots. Bellow is my config from asterisk 1.6 which worked, I`ve tried this in 1.8 and it’s not working. /features.conf/ // /[parkinglot_A] parkpos = 2011-2020 findslot = next parkingtime = 60 context = parked/ // /[parkinglot_B] parkpos = 2021-2030 findslot = next parkingtime = 60 context = parked / /[general]/ /parkext = 2000 / /parkpos = 2001-2110 / /parkingtime = 60 / /adsipark = yes / /findslot = next/ // // /sip.conf/ // /[user_A]/ /.../ /parkinglot=parkinglot_A/ // // /[user_B]/ /.../ /parkinglot=parkinglot_B/ // /extensions.conf/ // /include = parked/ // /exten = 2011,hint,park:2...@parked/ /exten = 2011,1,Wait(1)/ /exten = 2011,2,Set(CHANNEL(parkinglot)=parkinglot_A/ Is it just a typo that fact that you are missing a closing bracket on the above line (and two more like it below)? Sebastian /exten = 2011,3,ParkedCall(2011)/ // /exten = 2012,hint,park:2...@parked/ /exten = 2012,1,Wait(1)/ /exten = 2012,2,Set(CHANNEL(parkinglot)=parkinglot_A/ /exten = 2012,3,ParkedCall(2012)/ // /exten = 2021,hint,park:2...@parked/ /exten = 2021,1,Wait(1)/ /exten = 2021,2,Set(CHANNEL(parkinglot)=parkinglot_B/ /exten = 2021,3,ParkedCall(2021)/ // /exten = 2022,hint,park:2...@parked/ /exten = 2022,1,Wait(1)/ /exten = 2022,2,Set(CHANNEL(parkinglot)=parkinglot_B)/ /exten = 2022,3,ParkedCall(2022)/ With this config I was able to also monitor every parking place and see on the phone of there is a call parked on that extension and pick that up from every phone in the office and every user when they press “Park Call” are parking the current call to their parking lot. This config worked on asterisk 1.6 and we cannot make it to work on asterisk 1.8. Any ideas guys ? Thanks for the answers, Bogdan Sarandan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Multiple Parking Lots
Yes, just a typo, sorry for that, maybe I've deleted them when I`ve copy/pasted . Thanks, Bogdan -Original Message- From: Sebastian Sent: Monday, November 08, 2010 3:24 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 Multiple Parking Lots Hi, On 11/08/2010 08:59 PM, Bogdan Sarandan wrote: Hello, Recently we have been using asterisk 1.6 and everything worked ok, all our productions server are 1.6. Recently we have upgraded one to 1.8 and multiple parking call are not working, which worked pretty ok on asterisk 1.6. Are there any major changes in asterisk 1.8 related to park call and multiple parking lots. Bellow is my config from asterisk 1.6 which worked, I`ve tried this in 1.8 and it’s not working. /features.conf/ // /[parkinglot_A] parkpos = 2011-2020 findslot = next parkingtime = 60 context = parked/ // /[parkinglot_B] parkpos = 2021-2030 findslot = next parkingtime = 60 context = parked / /[general]/ /parkext = 2000 / /parkpos = 2001-2110 / /parkingtime = 60 / /adsipark = yes / /findslot = next/ // // /sip.conf/ // /[user_A]/ /.../ /parkinglot=parkinglot_A/ // // /[user_B]/ /.../ /parkinglot=parkinglot_B/ // /extensions.conf/ // /include = parked/ // /exten = 2011,hint,park:2...@parked/ /exten = 2011,1,Wait(1)/ /exten = 2011,2,Set(CHANNEL(parkinglot)=parkinglot_A/ Is it just a typo that fact that you are missing a closing bracket on the above line (and two more like it below)? Sebastian /exten = 2011,3,ParkedCall(2011)/ // /exten = 2012,hint,park:2...@parked/ /exten = 2012,1,Wait(1)/ /exten = 2012,2,Set(CHANNEL(parkinglot)=parkinglot_A/ /exten = 2012,3,ParkedCall(2012)/ // /exten = 2021,hint,park:2...@parked/ /exten = 2021,1,Wait(1)/ /exten = 2021,2,Set(CHANNEL(parkinglot)=parkinglot_B/ /exten = 2021,3,ParkedCall(2021)/ // /exten = 2022,hint,park:2...@parked/ /exten = 2022,1,Wait(1)/ /exten = 2022,2,Set(CHANNEL(parkinglot)=parkinglot_B)/ /exten = 2022,3,ParkedCall(2022)/ With this config I was able to also monitor every parking place and see on the phone of there is a call parked on that extension and pick that up from every phone in the office and every user when they press “Park Call” are parking the current call to their parking lot. This config worked on asterisk 1.6 and we cannot make it to work on asterisk 1.8. Any ideas guys ? Thanks for the answers, Bogdan Sarandan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Addons for Asterisk 1.8?
I just noticed that there is no Addons package for 1.8, does that mean that I can use asterisk-addons-1.6.2.2 with Asterisk 1.8? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addons for Asterisk 1.8?
The addons are in the same package. Regards - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: Asterisk asterisk-users@lists.digium.com Sent: Monday, November 08, 2010 4:43 PM Subject: [asterisk-users] Addons for Asterisk 1.8? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addons for Asterisk 1.8?
On Mon, 2010-11-08 at 16:53 -0500, bakko wrote: The addons are in the same package. Regards - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: Asterisk asterisk-users@lists.digium.com Sent: Monday, November 08, 2010 4:43 PM Subject: [asterisk-users] Addons for Asterisk 1.8? Yes, I spoke before opening the UPGRADE.txt file and reading that they are now included in the same archive. I do not quite understand why they changed the distribution method. I think it is better to have a separate package that you do not have to download every time you upgrade Asterisk. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Addons for Asterisk 1.8?
The addons for 1.8 are included in asterisk, look at the menu in make menuselect while compiling asterisk. On Mon, Nov 8, 2010 at 10:43 PM, Carlos Chavez cur...@telecomabmex.comwrote: I just noticed that there is no Addons package for 1.8, does that mean that I can use asterisk-addons-1.6.2.2 with Asterisk 1.8? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Ondrej Škopek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
On Mon, Nov 8, 2010 at 7:05 AM, Cary Fitch ca...@usawide.net wrote: Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed bit? It won't be ISDN. It will be some form of RBS. You probably have several choices as to which type of RBS (probably several ESF options, you'll probably pick one of them; you may be able to use SF as well). You should probably work with your LEC to figure out exactly what they will hand off to you. You might make a costly mistake if you don't. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Thanks. I think I would still need a firewall. Maybe a 1u rack double enclosure for two Alix boards - one as firewall - and one as PBX would do better. Anyhow, I don't want to open the box if I don't have to. Is there any way I can push the .gz file over console cable rather than putting the CF in a reader? Thanks On Mon, Nov 8, 2010 at 1:06 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
Hello, I got the time to upgrade my tutorial about Asterisk and Kamailio realtime integration to latest stable release of Kamailio, version 3.1.0 (out on Oct 6, 2010). You can find the document at: * http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb Besides making it work for v3.1.x, the Kamailio config file has some new features included: * IP authentication - can be enabled via define WITH_IPAUTH * TLS support - can be enabled via define WITH_TLS - TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure Asterisk on UDP * detection of DoS attacks - can be enabled via define WITH_ANTIFLOOD - banning automatically traffic from attacker IP addresses for a specific time interval * restructuring of configuration file for better modularity and highlighting of functionalities such as registrar, location server, within-dialog request routing Hope it is useful for some people within this community. Next step, naturally, is to upgrade the tutorial for latest Asterisk, 1.8.0, just needs some time to get familiar with it. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound call issue...
Not sure if you read the configs I attached, but that line is already in there... Guess again... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, November 03, 2010 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] inbound call issue... insecure=very should fix it. On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack gmals...@gmellc.com wrote: Can anyone tell me why my inbound calls keep getting rejected with 401? Here’s the debug information: --- SIP read from UDP:147.135.32.221:5060 --- INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact: sip:4144038...@147.135.32.221:5060 Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.248 t=0 0 m=audio 15502 RTP/AVP 0 18 8 96 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 - [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- (11 headers 14 lines) --- [Nov 3 02:08:40] VERBOSE[7207] netsock.c: == Using SIP RTP CoS mark 5 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 (NAT) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis request - 31007e...@147.135.32.221 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for '4144038968' from 147.135.32.221:5060 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- Reliably Transmitting (NAT) to 147.135.32.221:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281-;received=147.135.32.221 From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22;tag=as4fffe111 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2dd58be8 Content-Length: 0 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Scheduling destruction of SIP dialog '31007e...@147.135.32.221' in 32000 ms (Method: INVITE) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- SIP read from UDP:147.135.32.221:5060 --- ACK sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 ACK From: Wi Msip:number f...@147.135.32.221;user=phone;tag=9bbc To: usernamesip:s...@216.26.109.22;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length:0 Here’s the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register = 6087294351:sip password@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=6087294351 secret=sip password username=6087294351 insecure=very context=DID_trunk_1 authname=6087294351 dtmfmode=inband dtmf=inband canreinvite=no [guest] type=friend host=dynamic canreinvite=no context=DID_trunk_1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] inbound call issue...
You didn't say which version of Asterisk you were using. insecure=very is deprecated in favor of insecure=port,invite Many of the VoIP providers do not have this right in their examples. Darrick On 11/08/2010 05:52 PM, Gregory Malsack wrote: Not sure if you read the configs I attached, but that line is already in there... Guess again... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, November 03, 2010 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] inbound call issue... insecure=very should fix it. On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsackgmals...@gmellc.com wrote: Can anyone tell me why my inbound calls keep getting rejected with 401? Here’s the debug information: --- SIP read from UDP:147.135.32.221:5060 --- INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact:sip:4144038...@147.135.32.221:5060 Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.248 t=0 0 m=audio 15502 RTP/AVP 0 18 8 96 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 - [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- (11 headers 14 lines) --- [Nov 3 02:08:40] VERBOSE[7207] netsock.c: == Using SIP RTP CoS mark 5 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 (NAT) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis request - 31007e...@147.135.32.221 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for '4144038968' from 147.135.32.221:5060 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- Reliably Transmitting (NAT) to 147.135.32.221:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281-;received=147.135.32.221 From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22;tag=as4fffe111 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2dd58be8 Content-Length: 0 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Scheduling destruction of SIP dialog '31007e...@147.135.32.221' in 32000 ms (Method: INVITE) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- SIP read from UDP:147.135.32.221:5060 --- ACK sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 ACK From: Wi Msip:number f...@147.135.32.221;user=phone;tag=9bbc To: usernamesip:s...@216.26.109.22;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length:0 Here’s the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register = 6087294351:sip password@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=6087294351 secret=sip password username=6087294351 insecure=very context=DID_trunk_1 authname=6087294351 dtmfmode=inband dtmf=inband canreinvite=no [guest] type=friend host=dynamic canreinvite=no context=DID_trunk_1 -- Darrick Hartman DJH Solutions, LLC
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Bruce B wrote: Thanks. I think I would still need a firewall. Maybe a 1u rack double enclosure for two Alix boards - one as firewall - and one as PBX would do better. Anyhow, I don't want to open the box if I don't have to. Is there any way I can push the .gz file over console cable rather than putting the CF in a reader? DO you mean once you have built the system? AstLinux has an upgrade facility built into the system, with the ability to revert to the previous version, all built into the web interface. To initially build the system, it seems to me you would need to put the first OS on the CF card to get the board alive. And if you have an Alix with 2 Ethernet ports, why a second one as a firewall? AstLinux has a built in firewall You did say a SMALL office, didn't you? John Novack Thanks On Mon, Nov 8, 2010 at 1:06 PM, Darrick Hartman (lists) dhart...@djhsolutions.com mailto:dhart...@djhsolutions.com wrote: Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Yes, it is a small office. I am familiar with pfSense. I am not sure if firewall on Astlinux is as versatile and flexible. But also, I am wondering if with all those attacks around now-a-days if the box will be able to handle 5 extensions, voicemail, IVR, firewall, DHCP, openvpn all together. Thanks On Mon, Nov 8, 2010 at 7:24 PM, John Novack jnov...@stromberg-carlson.orgwrote: Bruce B wrote: Thanks. I think I would still need a firewall. Maybe a 1u rack double enclosure for two Alix boards - one as firewall - and one as PBX would do better. Anyhow, I don't want to open the box if I don't have to. Is there any way I can push the .gz file over console cable rather than putting the CF in a reader? DO you mean once you have built the system? AstLinux has an upgrade facility built into the system, with the ability to revert to the previous version, all built into the web interface. To initially build the system, it seems to me you would need to put the first OS on the CF card to get the board alive. And if you have an Alix with 2 Ethernet ports, why a second one as a firewall? AstLinux has a built in firewall You did say a SMALL office, didn't you? John Novack Thanks On Mon, Nov 8, 2010 at 1:06 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is this a DDoS to reach Asterisk?
Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 3. 006367 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 20. 274886 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 2. 893859 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 28. 739620 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 2. 936286 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 1. 207250 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 3. 015370 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 7. 088359 rule 67/0(match): block in on vr1: 61.130.103.10 69.90.78.42: [|icmp] 11. 825521 rule 67/0(match): block in on vr1: 71.100.221.211.4521 69.90.78.33.445: [|tcp] 2. 316564 rule 67/0(match): block in on
[asterisk-users] Store CDR (call detail record) to Oracle database
Hi all, Now i want to store cdr (call detail record) to Oracle database but i don't know how to do .Can anyone help me ? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is this a DDoS to reach Asterisk?
Bruce B wrote: Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 3. 006367 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 20. 274886 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 2. 893859 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 28. 739620 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 2. 936286 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 1. 207250 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 3. 015370 rule 67/0(match): block in on vr1: 118.171.176.188.42965 69.90.78.43.445: [|tcp] 7. 088359 rule 67/0(match): block in on vr1: 61.130.103.10 69.90.78.42 http://69.90.78.42: [|icmp] 11.
Re: [asterisk-users] Is this a DDoS to reach Asterisk?
And that's the problem. There is no such service running or such port is not open. They only keep trying this for no reason. It might cost us bandwidth for no reason. In fact there is no open ports on our network whatsoever. Thanks On Mon, Nov 8, 2010 at 9:50 PM, Lyle Giese l...@lcrcomputer.net wrote: Bruce B wrote: Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 3. 006367 rule 67/0(match): block in on vr1: 77.34.234.241.1899 69.90.78.43.445: [|tcp] 20. 274886 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 2. 893859 rule 67/0(match): block in on vr1: 66.211.120.62.1132 69.90.78.55.445: [|tcp] 28. 739620 rule 67/0(match): block in on vr1: 117.197.247.151.1042 69.90.78.55.445: [|tcp] 2. 936286 rule 67/0(match): block in on vr1: 117.197.247.151.1042
Re: [asterisk-users] Is this a DDoS to reach Asterisk?
Welcome to the Internet! It's a fact of life when having equipment connected to the Internet. The script kiddies are always probing and trying. Lyle Bruce B wrote: And that's the problem. There is no such service running or such port is not open. They only keep trying this for no reason. It might cost us bandwidth for no reason. In fact there is no open ports on our network whatsoever. Thanks On Mon, Nov 8, 2010 at 9:50 PM, Lyle Giese l...@lcrcomputer.net mailto:l...@lcrcomputer.net wrote: Bruce B wrote: Hi Everyone, I have pfSense running which supplies Asterisk with DHCP. I had some testing ports opened for a web server which I have totally closed now but when I chose option 10 (filter log) on pfSense I get all of this type of traffic (note that it was only 1 single IP and once I blocked that one it was like opening a can full of bees with all different IPs): tcpdump: WARNING: pflog0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on pflog0, link-type PFLOG (OpenBSD pflog file), capture size 96 bytes 00 rule 70/0(match): block in on vr1: 221.132.34.165.33556 69.90.78.53.52229: tcp 20 [bad hdr length 0 - too short, 20] 6. 239658 rule 70/0(match): block in on vr1: 121.207.254.227.6667 69.90.78.38.3072: tcp 24 [bad hdr length 0 - too short, 20] 7. 986724 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 867707 rule 70/0(match): block in on vr1: 61.231.237.223.4155 69.90.78.62.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 799337 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 931814 rule 70/0(match): block in on vr1: 186.36.73.212.4545 69.90.78.56.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 574556 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 956066 rule 70/0(match): block in on vr1: 190.7.59.45.1341 69.90.78.43.445: tcp 28 [bad hdr length 0 - too short, 20] 1. 598334 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 20 [bad hdr length 8 - too short, 20] 072759 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 109451 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 28 [bad hdr length 0 - too short, 20] 2. 731065 rule 70/0(match): block in on vr1: 2.95.19.121.3463 69.90.78.42.445: tcp 16 [bad hdr length 12 - too short, 20] 159413 rule 70/0(match): block in on vr1: 123.192.177.2.54518 69.90.78.43.445: tcp 20 [bad hdr length 8 - too short, 20] 374293 rule 70/0(match): block in on vr1: 219.163.19.138.3723 69.90.78.63.445: tcp 16 [bad hdr length 12 - too short, 20] 10. 234202 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 2. 985558 rule 70/0(match): block in on vr1: 189.105.69.200.2413 69.90.78.52.445: tcp 20 [bad hdr length 12 - too short, 20] 13. 236084 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 2. 982122 rule 70/0(match): block in on vr1: 82.51.36.230.2923 69.90.78.35.445: tcp 16 [bad hdr length 12 - too short, 20] 18. 493312 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 2. 477084 rule 70/0(match): block in on vr1: 218.16.118.242.80 69.90.78.47.39781: tcp 16 [bad hdr length 12 - too short, 20] 9. 92 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: tcp 16 [bad hdr length 12 - too short, 20] 1. 216002 rule 70/0(match): block in on vr1: 172.168.0.4.1568 69.90.78.49.445: [|tcp] 321600 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: tcp 20 [bad hdr length 8 - too short, 20] 1. 383839 rule 70/0(match): block in on vr1: 121.243.16.214.1677 69.90.78.54.445: [|tcp] 1. 466115 rule 70/0(match): block in on vr1: 72.179.18.165.2854 69.90.78.55.445: [|tcp] 7. 977140 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 2. 920013 rule 70/0(match): block in on vr1: 41.72.209.67.4532 69.90.78.36.445: [|tcp] 29. 032839 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 2. 996906 rule 70/0(match): block in on vr1: 201.168.49.13.1404 69.90.78.55.445: [|tcp] 62. 079279 rule 70/0(match): block in on vr1: 82.165.131.28.6005 69.90.78.47.1024: [|tcp] 34. 224871 rule 67/0(match): block in on
