[asterisk-users] All incoming calls landing in [customers] context
Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: = [default] include = dial_GUEST [customers] include = parkedcalls include = dial = The contexts, dial, and dial_GUEST essentially handle all call routing, with the idea that guests (anonymous internet callers) can't get out to the pstn. The problem is that ALL incoming calls are landing in [customers] even if the caller is an unregistered SIP client. As soon as a call comes in, I see it jump immediately to x...@customers:1 and this happends with registered or unregistered clients. What am I doing wrong? -- Take care and have fun, Mike Diehl. -- _ -- 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] Being attacked by an Amazon EC2
I worked with Project Honeypot guys for a while, they are more than willing to assist, as they already have the backend work done for a clearing house identifying hackers. The biggest issue we had a year ago was to create the mechanism in asterisk to push valid log messages out to the database and then determine what to do with that data? Because I run a lot of forums and blogs, I use Project Honeypot, report to them and have lent them a few honeypot MX and pages. I tried to bridge the gap between a few Asterisk developers and the Honeypot developers, ultimately the project stalled and I got busy with other matters. If anyone here would like to pick up the torch and move this along, I can certainly provide info on how far along we got and contact info for the parties involved. Project Honeypot seems pretty overworked/overstretched already, but if you're able to communicate whith them that's excellent, they are doing a great job with their DB, it saves me a lot of time. Please contact me if you have time to work on this and are interested. I'm sure the Project Honeypot guys will be willing to pick this project back up and work on it. I can't contribute code, but I can help spread the word. I also still believe that Amazon needs to put resources to work on the problem. The cloud is too easy to hide in for what are obviously fraudulent operations. We will certainly be talking about this on the VoIP Users Conference in the next weeks. We should schedule it as a topic, possibly for the April 30th. Would you be available for that JR? (12 Noon EDT) /r -- _ -- 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] Being attacked by an Amazon EC2 ...
On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman dhart...@djhsolutions.com wrote: That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. With the growth of the cloud offerings, this problem will likely grow, so yes, a generic solution is needed. What I want to see though, and no provder has done much if anything about it, is REPORTING and INVESTIGATION. It is easy to use a script to report and submit, we can all do that, even I could (if I had a box running and needed to). The hard part is them having their tech/sys people actually look at the network and see, Oh, ya, there's some shit happening that on that instance... If Amazon's form submit didn't even work, that's a really bad reflection on their brand in a lot of ways, including tech competence. If that is know to geeks like us, it won't hurt them which is why, like a broken record, I keep saying: put your Amazon experience out to the public. When it starts being mentioned in Wired, Storm Cloud or something, THEN Amazon will have to do something. I do not believe Amazon is taking reasonable measures now in doing their job, and that they should be working towards that goal, reasonable measures as opposed to NO measures. /r -- _ -- 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 room monitor
On Mon, Apr 12, 2010 at 8:19 PM, Mark Hulber asterisk.ad...@hulber.com wrote:
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
A phone with a good speaker phone.
Ability to make the audio one way. I want to monitor the room but not
have my voice heard in the room. Yes, the mute button can accomplish
this also.
I have been using the SPA942's around the house (the speaker is just ok
but probably good enough). Can I set one of these or a similar Cisco
phone to auto answer in speaker mode? Any ideas on an alternative phone
that would allow this?
The alternative is to just set up the call locally and then leave the
room with the line open but ideally I'd like to be able to open up the
monitor on demand.
Thanks,
MARK.
Hello Mark,
Please find bellow a dialplan proof-of-concept for your requirement
(is based on intercom module present in FreePBX and adapted to have
only one way audio for 60 secconds). We have tested with Linksys
SPA9XX phones and works fine (hint: clear regional=call progres
tones=page tone in order to cancel the page tone if you need to be
super-silent).
HTH,
Ioan Indreias
www.modulo.ro
exten = _6XX,1,Answer
exten = _6XX,n,Set(_ALERTINFO=Alert-Info: Ring Answer)
exten = _6XX,n,Set(_CALLINFO=Call-Info: uri\;answer-after=0)
exten = _6XX,n,Set(_SIP_URI_OPTIONS=intercom=true)
exten = _6xx,n,SipAddHeader,${ALERTINFO})
exten = _6XX,n,SipAddHeader,${CALLINFO})
exten = _6XX,n,Dial(SIP/1${EXTEN:1},5,G(100))
exten = _6XX,100,Goto(200)
exten = _6XX,101,Goto(300)
exten = _6XX,200,ChanSpy(SIP/1${EXTEN:1})
exten = _6XX,300,Wait(60)
exten = _6XX,n,Hangup
--
_
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Re: [asterisk-users] Being attacked by an Amazon EC2 ...
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'm not aware of the asterisk.dev list but maybe someone can tell if they can help us here? Alyed 2010/4/13 Randy R randulo2...@gmail.com On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman dhart...@djhsolutions.com wrote: That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. With the growth of the cloud offerings, this problem will likely grow, so yes, a generic solution is needed. What I want to see though, and no provder has done much if anything about it, is REPORTING and INVESTIGATION. It is easy to use a script to report and submit, we can all do that, even I could (if I had a box running and needed to). The hard part is them having their tech/sys people actually look at the network and see, Oh, ya, there's some shit happening that on that instance... If Amazon's form submit didn't even work, that's a really bad reflection on their brand in a lot of ways, including tech competence. If that is know to geeks like us, it won't hurt them which is why, like a broken record, I keep saying: put your Amazon experience out to the public. When it starts being mentioned in Wired, Storm Cloud or something, THEN Amazon will have to do something. I do not believe Amazon is taking reasonable measures now in doing their job, and that they should be working towards that goal, reasonable measures as opposed to NO measures. /r -- _ -- 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] Dahdi, junghanns and qozap
2010/4/12 Olivier oza_4...@yahoo.fr 2010/4/12 Olivier oza_4...@yahoo.fr Hi, In my 1.6.1.18 with dahdi 2.2.1.1, I've got : # dahdi_hardware pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card Does it mean I should download and use qozap or is it a bug in Dahdi ? Regards I should have added that I'm using an old Junghanns OctoBRI ... Regards Hi, I gave another shot using a Junghanns PCI-E QuadBRI. Reading https://issues.asterisk.org/view.php?id=16447, I installed trunk (revision 8519) for dahdi-linux and dahdi-tools. Result is: # dahdi_hardware pci::06:04.0 qozap- 1397:08b4 Generic Cologne ISDN card If I'm not mistaken, it seems this QuadBRI board is still associated (man dahdi_hardware) with a qozap driver which is currently not loaded. Am I correct to think I should hopefully see something like : # dahdi_hardware pci::06:04.0 wcb4xxp+ 1397:08b4 Generic Cologne ISDN card 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] All incoming calls landing in [customers] context
Have a look at: http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication It's about IAX but guess will give you some good hints on how to solve your problem. Alyed 2010/4/13 Mike Diehl mdi...@diehlnet.com Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: = [default] include = dial_GUEST [customers] include = parkedcalls include = dial = The contexts, dial, and dial_GUEST essentially handle all call routing, with the idea that guests (anonymous internet callers) can't get out to the pstn. The problem is that ALL incoming calls are landing in [customers] even if the caller is an unregistered SIP client. As soon as a call comes in, I see it jump immediately to x...@customers:1 and this happends with registered or unregistered clients. What am I doing wrong? -- Take care and have fun, Mike Diehl. -- _ -- 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] Flood of REGISTERs - attack?
On 13/04/10 00:27, Tom Stordy-Allison wrote: Yep - this is the same codebase - the attack that I had from an EC2 yesterday and the day before, all had the User-Agent: friendly-scanner too. Looks like they are branching out Go with Joshua Steins blog post - it worked perfect for me and got it off my back. Unfortunately, it hasn't worked here. Took me ages to figure why iptables -t nat -A PREROUTING -i ppp0 -s 62.149.239.97 -p udp --dport 5060 -j REDIRECT --to-port 5071 didn't redirect the traffic. Turns out (I think) that only new connections are sent to the nat table, and this ones been established for several days now. If anyone can shed light on how to reset the connection tracking I'd be interested, but only academically now. Instead I just stopped asterisk and ran Joshua Stein's script on 5060. But it didn't do the trick. The bot showed no sign whatsoever of letting up. My other line of defence is the following rate limiting in iptables. Is this likely to interfere with actual day to day operations of Asterisk, given a small and not very busy installation? Basically it will drop packets if it has seen more than 20 in the last 30 seconds, or more than 10 in the last 2 seconds from the same host. # rate limit external SIP connections to Asterisk iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --rcheck --seconds 30 --hitcount 20 -m limit --limit 1/sec --limit-burst 3 -j LOG --log-prefix Dropped (sip rate lim 1): iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --update --seconds 30 --hitcount 20 -j DROP iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --rcheck --seconds 2 --hitcount 10 -m limit --limit 1/sec --limit-burst 3 -j LOG --log-prefix Dropped (sip rate lim 2): iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --update --seconds 2 --hitcount 10 -j DROP iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --set -- _ -- 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] Dahdi, junghanns and qozap
On Mon, Apr 12, 2010 at 06:16:51PM +0200, Olivier wrote: Hi, In my 1.6.1.18 with dahdi 2.2.1.1, I've got : # dahdi_hardware pci::01:0a.0 qozap- 1397:16b8 Junghanns OctoBRI ISDN card Does it mean I should download and use qozap or is it a bug in Dahdi ? DAHDI 2.3.0 includes support for more such cards. As you can see, the difference is mostly adding extra IDs. http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/wcb4xxp_extra_trunk?view=markup http://svn.debian.org/viewsvn/pkg-voip/dahdi-tools/trunk/debian/patches/wcb4xxp_extra_trunk?view=markup -- 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
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On Tue, Apr 13, 2010 at 08:27:11AM +0200, Randy R wrote: On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman dhart...@djhsolutions.com wrote: That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. With the growth of the cloud offerings, this problem will likely grow, so yes, a generic solution is needed. What I want to see though, and no provder has done much if anything about it, is REPORTING and INVESTIGATION. It is easy to use a script to report and submit, we can all do that, even I could (if I had a box running and needed to). The hard part is them having their tech/sys people actually look at the network and see, Oh, ya, there's some shit happening that on that instance... But this potentially moved DoS attacks from one place to another. Especially given that the source of a UDP packet is easy to forge. (And yes, in this case the attack was not intended to be a simple DoS) -- 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
Re: [asterisk-users] Being attacked by an Amazon EC2
On Mon, Apr 12, 2010 at 04:58:42PM -0500, JR Richardson wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ This is for really bad spammers. In our case it would be used to block Amazon AWS in the (completely unlikely!) case that they would do nothing about those cases. We could establish something similar to that for VOIP attacks. It may not be exactly a trivial system to maintain such a list. (removing IP's after X amount of time, disputing false claims etc). Maybe someone could contact spamhaus to create a list for VOIP since they seem to have a nice system in place? Hi All, good discussion, similar to ones we had a year or so ago. The RBL concept is valid, at least to get a repository going that list malicious activity specific to SIP attacks. n I worked with Project Honeypot guys for a while, they are more than willing to assist, as they already have the backend work done for a clearing house identifying hackers. The biggest issue we had a year ago was to create the mechanism in asterisk to push valid log messages out to the database and then determine what to do with that data? I tried to bridge the gap between a few Asterisk developers and the Honeypot developers, ultimately the project stalled and I got busy with other matters. If anyone here would like to pick up the torch and move this along, I can certainly provide info on how far along we got and contact info for the parties involved. Please contact me if you have time to work on this and are interested. I'm sure the Project Honeypot guys will be willing to pick this project back up and work on it. I've been bitten too many times by over-jelous anti-spam black lists. It's easy to get in. More difficult to be removed. And heck, I can easily get set up a few servers in Amazon which will generate faked logs of attacks from your server, if I want to shut your phone system for a couple of days. -- 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
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
- Original Message - Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'm not aware of the asterisk.dev list but maybe someone can tell if they can help us here? Alyed 2010/4/13 Randy R randulo2...@gmail.com On Mon, Apr 12, 2010 at 7:17 PM, Darrick Hartman dhart...@djhsolutions.com wrote: That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. With the growth of the cloud offerings, this problem will likely grow, so yes, a generic solution is needed. What I want to see though, and no provder has done much if anything about it, is REPORTING and INVESTIGATION. It is easy to use a script to report and submit, we can all do that, even I could (if I had a box running and needed to). The hard part is them having their tech/sys people actually look at the network and see, Oh, ya, there's some shit happening that on that instance... If Amazon's form submit didn't even work, that's a really bad reflection on their brand in a lot of ways, including tech competence. If that is know to geeks like us, it won't hurt them which is why, like a broken record, I keep saying: put your Amazon experience out to the public. When it starts being mentioned in Wired, Storm Cloud or something, THEN Amazon will have to do something. I do not believe Amazon is taking reasonable measures now in doing their job, and that they should be working towards that goal, reasonable measures as opposed to NO measures. /r DNS lookup capability appears to be required on a Asterisk installation and hence a DNSRBL would appear to be a good solution. A alternative, similar to the SaneSecurity AV sigs, would be to have a pool of rsync servers for downloading a list of known IPs. Again this would require community contribution in both time and resources. I would be happy to allocate some spare memory and CPU cycles and hopefully my employer would as-well. -- Thanks, Phil -- _ -- 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] Being attacked by an Amazon EC2 ...
On Tue, 13 Apr 2010, Alyed wrote: Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) The biggest issue I see is that people are installing Asterisk and other high-level applications on top of Linux (and other *nix'es) without the experience of sysadmin - then when something goes wrong they want the application to fix it rather than apply some basic and pretty fundamental sysadmin techniques to solve the issue. And that means that even having permit= and deny= in sip.conf and iax.conf, etc. is too much. With proper OS level firewalling they're simply not needed and do nothing more than add another potential point of failure and add yet more code to maintain. Gordon -- _ -- 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] SNOM M9 base station A to base station B
Hello, I have a question concerning SNOM M9 base station. If my customer places a SNOM M9 base station in place A and a SNOM M9 base station in place B, which is 100 meters further... will a SNOM M9 handheld from base station A register to base station B when it enters its DECT-environment. Can one transparently walk from place A to place B with the same M9 handheld and not loose the conversation ?? Greetingz, 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
Re: [asterisk-users] SNOM M9 base station A to base station B
Have you tried the SNOM forum ? They would probably have more info for you http://forum.snom.com/ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 13 April 2010 10:12 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SNOM M9 base station A to base station B Hello, I have a question concerning SNOM M9 base station. If my customer places a SNOM M9 base station in place A and a SNOM M9 base station in place B, which is 100 meters further... will a SNOM M9 handheld from base station A register to base station B when it enters its DECT-environment. Can one transparently walk from place A to place B with the same M9 handheld and not loose the conversation ?? Greetingz, 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
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
- Original Message - On Tue, 13 Apr 2010, Alyed wrote: Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) The biggest issue I see is that people are installing Asterisk and other high-level applications on top of Linux (and other *nix'es) without the experience of sysadmin - then when something goes wrong they want the application to fix it rather than apply some basic and pretty fundamental sysadmin techniques to solve the issue. And that means that even having permit= and deny= in sip.conf and iax.conf, etc. is too much. With proper OS level firewalling they're simply not needed and do nothing more than add another potential point of failure and add yet more code to maintain. Gordon Gordon, Completely agree with what you are saying though I believe the proposal of some sort of shared IP list is a valid one. If you had not brought this to the attention of the list then this discussion would have not taken place. I am guilty in that when a EC2 server attempted to break into my PBX I did not share it with the list. We, large assumption, are all at some point subjected to probing attacks against our Asterisk deployments and I feel it would be great if there was some mechanism where we were able to share those hackers IPs for blocking by one means or another. -- Thanks, Phil -- _ -- 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] problem of when memory become 50% or more then sound become noisy?
Dear all,
Currently I am using asterisk 1.4.23.1. . Over the period of 1 week,
the memory in use starts off at 50% and
continues to climb until it hits 99%. When memory usage ratio become
50% or more, the quality of calls become
extremely noisy. The call quality goes back to being perfect once I
reboot the machine,
but I was to try and avoid having to reboot the machine every week.
the following is the memory status during the usage ratio of memory
approx. 50% or more which is in /proc/{process id of asterisk}/smaps
file :
09001000-7ebf6000 rw-p 09001000 00:00 0 [heap]
Size: 1929172 kB
Rss:1679436 kB
Shared_Clean: 0 kB
Shared_Dirty: 0 kB
Private_Clean: 6472 kB
Private_Dirty: 1672964 kB
Swap: 224172 kB
My server's RAM size is:
MemTotal: 4147888 kB
Processor is :
Intel(R) Xeon(R) CPU X5460 @ 3.16GHz
Asterisk and the User-Agent is connected through the Internet.
Is it the prablem of memory leakage of asterisk? Is there any solution
to solve this memory's problem?
is it asterisk's bug or something else? I cannot find out the solution
and cannot find out where is the problem?
Presently, I need this solution very urgently. I am eagerly waiting for reply.
Or is there any solution to clean up the memory's usage space in
asterisk source code?
Thanks in advance
Nahar
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Re: [asterisk-users] problem of when memory become 50% or more then sound become noisy?
On Tue, Apr 13, 2010 at 06:42:50PM +0900, kamrun nahar bina wrote: Dear all, Currently I am using asterisk 1.4.23.1. . Over the period of 1 week, the memory in use starts off at 50% Is there much active swapping? Run 'vmstat 1' for a while. Look at the columns 'si' (swap in) and so (swap out). -- 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
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
Am 13.04.2010 10:47, schrieb Gordon Henderson: I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) The biggest issue I see is that people are installing Asterisk and other high-level applications on top of Linux (and other *nix'es) without the experience of sysadmin - then when something goes wrong they want the application to fix it rather than apply some basic and pretty fundamental sysadmin techniques to solve the issue. And that means that even having permit= and deny= in sip.conf and iax.conf, etc. is too much. With proper OS level firewalling they're simply not needed and do nothing more than add another potential point of failure and add yet more code to maintain. Gordon I definitely do to agree with Gordon! If you have to get your car over a river, try to find a bridge or ferry instead of trying to teach the car swimming O.k., maybe this was a bit polemic. But in some way, it reminds me of Linux. What I really love ist the very high flexibility. And I definitely can see Gordon's point, not adding functionality to programs which somehow doesn't belong there. My thought is: It's very easy to write a program/script which connects to any random IP:port adress and sends packets there. Regardless if the remote side is responding or not. This way you can easily eat up the remote side's bandwith and/or data volume limit. And there's nothing the remote side can do against it except pulling the plug. If someone is sending millions of registers triyng to find an entry into a phone server, the problem is related to asterisk. But as soon as a firewall can block that, (or even as long as asterisk's security is strong enough to not let them in), the issue is NOT related to asterisk any more. From that moment on it is reduced to a bandwith eat-up problem and belongs to the area of network administration. This moves into the direction of an academic discussion titled what can I do if someone else eats up my bandwith/data-volume-limit? My 2 cents.. BTW, the good news: had no attack here within the last 48 hours. I implemented the iptables rules to drop packets from various adress ranges. But log them first. I'd like to see if the bot is continuing if it doen't get any reponses or if it gives up. But no attack so far Norbert -- _ -- 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] problem of when memory become 50% or more then sound become noisy?
Dear Tzafrir Cohen, Now I executed vmstat 1, Now memory usage is 15% thats why (swap in) and so (swap out) is 0. But When memory usage become 50% or more then swap size become 224172 kB according to previous log. May be this is the reason for becoming sound noisy? But How i will solve this memory's problem of asterisk? Thanks in advance Nahar On Tue, Apr 13, 2010 at 7:04 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Apr 13, 2010 at 06:42:50PM +0900, kamrun nahar bina wrote: Dear all, Currently I am using asterisk 1.4.23.1. . Over the period of 1 week, the memory in use starts off at 50% Is there much active swapping? Run 'vmstat 1' for a while. Look at the columns 'si' (swap in) and so (swap out). -- 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] All incoming calls landing in [customers] context
You need to post your sip.conf and any included files in it. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-13 2:04 AM, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: = [default] include = dial_GUEST [customers] include = parkedcalls include = dial = The contexts, dial, and dial_GUEST essentially handle all call routing, with the idea that guests (anonymous internet callers) can't get out to the pstn. The problem is that ALL incoming calls are landing in [customers] even if the caller is an unregistered SIP client. As soon as a call comes in, I see it jump immediately to x...@customers:1 and this happends with registered or unregistered clients. What am I doing wrong? -- Take care and have fun, Mike Diehl. -- _ -- 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] protocol used to connect Asterisk and GSM core network (MSC)
Hi all, My Asterisk connect to GSM core network (connect directly to MSC) through E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ? Thanks in advance -- _ -- 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] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
This is not actually a problem... it's a side affect of how older versions of libpri handled PTMP links. Basically, after 3-5 minutes, the other side is probably trying to drop layers 1 and 2 due to no calls being active. For the most part, unless you see any issues, you should just ignore the message. This is just libpri re-establishing layer when the other side tries to drop it, due to its desire to have the perception of a persistent layer 2 (in older versions). In newer libpri (1.4 branch) it allows layer 2 to drop and stay dropped until it is needed by layer 3. Matthew Fredrickson Digium, Inc. Darshaka Pathirana wrote: Hi everyone. We have a problem here... Hope somebody can give us some hints. We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem. Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+lenny1) with zaptel (1.4.11) and libpri (1.4.3) is installed. There is a QuadBRI-Card installed: # lspci -vv -s 06:04.0 06:04.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Device b752 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Interrupt: pin A routed to IRQ 30 Region 0: I/O ports at cc00 [size=8] Region 1: Memory at fb6ff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- zttest gives me an average of 99.992% and zttool shows no alarms. But every about 3,5 minutes we get this (with debug span 1 enababled): 1 -- Timeout occured, restarting PRI 1 q921.c:859 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED 1 Sending Set Asynchronous Balanced Mode Extended 1 q921.c:534 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Apr 10 12:16:05] WARNING[28541]: chan_zap.c:2498 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! 1 Sending Set Asynchronous Balanced Mode Extended 1 -- Got UA from network peer Link up. 1 -- Restarting T203 counter == Primary D-Channel on span 1 up % cat /etc/zaptel.con # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: ztqoz/1/1 quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0) (MASTER) span=1,1,3,ccs,ami # termtype: te bchan=1-2 dchan=3 # Span 2: ztqoz/1/2 quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 0) span=2,2,0,ccs,ami # termtype: te bchan=4-5 dchan=6 # Span 3: ztqoz/1/3 quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 0) span=3,3,0,ccs,ami # termtype: te bchan=7-8 dchan=9 # Span 4: ztqoz/1/4 quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0) span=4,4,0,ccs,ami # termtype: te bchan=10-11 dchan=12 # Global data loadzone= at defaultzone = at % cat /etc/asterisk/zapata.conf [channels] language=de switchtype=euroisdn pridialplan=unknown prilocaldialplan=dynamic priindication=passthrough context=incoming immediate=no usecallingpres=yes usecallerid=yes group=1 nationalprefix=00 internationalprefix=000 signalling=bri_cpe echocancel=Yes overlapdial=Yes ; group=2 ; signalling=bri_cpe ; context=incoming ; channel = 10-11 ; channel = 1-2 ; channel = 4-5 ; channel = 7-8 ; channel = 10-11 (Only one span is connected to ISDN right now.) qozap is loaded and ztcfg -v gives me: Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) 12 channels to configure. Any idea what this could mean and how this could be fixed? Any help would be helpful. Thx. Greetings, - Darsha -- _ -- 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] Dial an extension with follow me
Hi people, I have an extension which has configured the follow me (it derives to an IVR). If in my dialplan I put Dial(extenX) (where extenX is that extension) and if it is not available, it should execute the IVR, is that right? Well, I think it should be, but it doesn't... Here is my CLI: Starting SIP/CALLUS-0b3f at join-dial,,1 failed so falling back to exten 's' -- Executing [...@join-dial:1] NoOp(SIP/CALLUS-0b3f, join-dial: START) in new stack -- Executing [...@join-dial:2] GotoIf(SIP/CALLUS-0b3f, 1?queue:conti) in new stack -- Goto (join-dial,s,3) -- Executing [...@join-dial:3] Gosub(SIP/CALLUS-0b3f, call-fm|s|1) in new stack -- Executing [...@call-fm:1] NoOp(SIP/CALLUS-0b3f, Start) in new stack -- Executing [...@call-fm:2] Dial(SIP/CALLUS-0b3f, SIP/3006) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/CALLUS-0b3f' status is 'CHANUNAVAIL' Thanks, Anahi Ludueña _ Aprende los trucos de Windows 7 con la gente que ya lo han probado Windows 7. http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- 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 Timezones
On 12 Apr 2010, at 22:14, asterisk-users-requ...@lists.digium.com wrote: There's system clock, and hardware clock. Whatever you get for the localtime when you do 'date' command is what you're going to get for logs from asterisk. It seems somewhere you have your system set to run in GMT, even though you don't want it to be like that. You will need to consult documentation about properly setting your clock for your timezone. The alternative is to leave your system 'broken', and change your time checks to GMT. Hello David, it was luckily easier than that! I checked with an extension in my DP that reads me the 'Asterisk time' and it was correct. At least after I tinkered a little with the time zones settings of the OS, but still with strange CDR times. So that I went looking into the CDR config file and noticed that I had the default choice of using GMT time for the CSV logs. So, no surprise that Asterisk was doing it... Thanks and best regards, Aldo -- _ -- 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] Being attacked by an Amazon EC2 ...
On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote: On Tue, 13 Apr 2010, Alyed wrote: Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) I'll agree with you here. Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) So a proper job for ip(6)tables, imho -- _ -- 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] PRI Gurus ONLY - Too complex of an issue
Thanks for the input. Problem was solved by adding transfer=no in zapata.conf For those who need TBCT, then add transfer=yes and facilityenable=yes in zapata.conf. However, if your telco has RLT or TBCT as a value added service and you have not subscribed to it then you will face my problem if transfer is not set to no -Bruce On Mon, Apr 12, 2010 at 11:28 PM, Don Kelly d...@donkelly.biz wrote: The symptoms look like you’re doing TBCT. Unless you’re recording or, for some other reason, want to supervise the call, TBCT is a more efficient use of your PRI as it frees up channels after the transfer. TBCT isn’t available with analog circuits, but is very similar to the analog flash and transfer. I started typing this a while ago and since see that you’re interested in call recording, so you don’t want TBCT. Good news is that you can indicate that you don’t want TBCT in your .conf files. Bad news is that I don’t know how you do it. But you’ve reduced the problem to its simplest form, and someone will respond with exactly what you need to do. And I see you figured out what it takes… --Don -- _ -- 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] dahdi_scan and OctoBRI. Bug or feature ?
Hi, When typing dahdi_scan on an OctoBRI-enabled setup, I've got only 8 replies such as : [1] active=yes alarms=RED ... [8] active=yes ... framing=CCS I would expect 16 replies (one per B-channel). Is this correct ? 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] SNOM M9 base station A to base station B
That is a function of the repeater. The repeater can manage 16 phones and pass them back to the base station. Always look at DECT as low power GSM because that is what it is... You have termination points and repeaters. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Apr 13, 2010 at 5:11 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I have a question concerning SNOM M9 base station. If my customer places a SNOM M9 base station in place A and a SNOM M9 base station in place B, which is 100 meters further... will a SNOM M9 handheld from base station A register to base station B when it enters its DECT-environment. Can one transparently walk from place A to place B with the same M9 handheld and not loose the conversation ?? Greetingz, 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 -- _ -- 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] cat /proc/zaptel/*
Hi all, On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is: ~# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A Zaptel Driver card 1 [TE] AMI/CCS 4 ZTHFC2/0/1 Clear 5 ZTHFC2/0/2 Clear 6 ZTHFC2/0/3 HDLCFCS These are two HFC-S PCI A cards. But, what exactly does all of this mean? In particular: * Span - In telephony, what is the definition of this term? * MASTER - How is this relevant? Only for timing purposes? * Clear - Is this said because only B-channels use ISDN clear codes? * HDLCFCS - Why say this about D-channels? Why not just say HDLC? * (In use) - What does this mean and how is this state determined? * 1 ZTHFC1/0/1 Clear (In use) - What do each of these columns specify? Thanks, Jaap -- _ -- 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] Being attacked by an Amazon EC2 ...
Hi! Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) However, I *still* think Asterisk should provide a delayreject option in sip.conf to greatly slow down answering request avanlanches. That will help to address the bandwidth issue if the attacker is configured to wait for a response before starting the next request. Apart from that here are the most important messages: Use strong passwords in sip.conf, and use keys in iax.conf, and avoid usernames that can be guessed too easily (numbers from 100 to and first names). Philipp -- _ -- 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] Full transfer details on inbound calls
Hi We're using asterisk 1.4.17 using RealTime and my boss has decided that we should keep a track of the full history of incoming calls i.e. who and when they were transferred to. The asterisk CDR only holds the initial answering channel for any call and not any further transfers that may have happened. The idea we are toying with is getting the time and the originating channel from the cdr, and then searching the full asterisk logs for the channel identifier string. Obviously we would have to have the verbose output going to a file and make sure that the verbosity in the console is always at least 5. I've done enough testing to see that is is possible i...@trinity:/var/log/asterisk$ grep 'Apr 13' full | grep SIP/xxx.xxx.xxx.xxx-082090e8 | grep answered [Apr 13 13:31:11] VERBOSE[17120] logger.c: -- SIP/811-08214f50 answered SIP/xxx.xxx.xxx.xxx-082090e8 [Apr 13 13:31:31] VERBOSE[17120] logger.c: -- SIP/808-08212f08 answered SIP/xxx.xxx.xxx.xxx-082090e8 The above output shows that the originating channel was answered by sip extension 811 and then by 808 20 seconds later. I am also considering parsing the full log into a mysql database and doing the searching in there. My question is is this a good way to go about what I'm trying to achieve or is there a simpler/less process intensive method that I'm missing. Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 users] asterisk realtime - database driven dialplan
i have installed the asterisk 1.6 before that installed the necessary
packages in Debian,
* i followed the steps as follows,
r...@astserver: ~# apt-get install unixodbc unixodbc-dev odbc-postgresql
postgresql-8.1 postgresql-contrib postgresql-dev
* then i installed the asterisk 1.6 version with the odbc modules as in the
selected list.
* then i created the database as asterisk and the user also with the same
name. The password is secret.
r...@astserver: ~# su postgres
$ createuser -s -D -R -l -P -e asterisk
$ createdb -O asterisk -e asterisk
* then i did the configuration as follows,
/etc/odbc.ini
[banking]
Description = ODBC Testing
Driver = PostgreSQL
Trace = No
Database= asterisk
Servername = 127.0.0.1
Username= asterisk
Password= secret
Port= 5432
/etc/odbcinst.ini
[PostgreSQL]
Description = ODBC for PostgresQL
Driver = /usr/lib/libodbcpsql.so
Setup = /usr/lib/odbc/libodbcpsqlS.so
FileUsage = 1
/etc/asterisk/res_odbc.conf -- following lines were included,
[postgres]
enabled = yes
dsn = banking
pre-connect = yes
/etc/asterisk/func_odbc.conf -- following lines were included,
[DUMMY]
dsn=postgres
read=select name from dialplan_data where ext='${SQL_ESC(${ARG1})}'
writesql=UPDATE dialplan_data SET name='${SQL_ESC(${VAL1})}' WHERE
ext='${SQL_ESC(${ARG1})}'
then i searched and there is no file available with name libodbcpsql.so.
How do can i make it possible to work ?
Thanks in Advance,
Balakrishnan M
r...@astserver: ~#isql -v banking
[IM004][unixODBC][Driver Manager]Driver's SQLAllocHandle on SQL_HANDLE_HENV
failed
[ISQL]ERROR: Could not SQLConnect
r...@astserver: ~#
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Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On Apr 13, 2010, at 8:04 AM, Hans Witvliet wrote: On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote: On Tue, 13 Apr 2010, Alyed wrote: Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) I'll agree with you here. Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) So a proper job for ip(6)tables, imho -- +1 for outside of asterisk. I want something that blocks it before it gets to the Asterisk processes. I've posted a little script on Team Forrest for how I'm blocking the traffic (using a quick perl script, iptables, and cron). The script is at http://bit.ly/cDHlLq ---fred http://qxork.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] Is restart of span a concern on PRI?
Hi Guys, I have been checking logs and noticed this over the last night. Should I be concerned? and where to look for further details? Sample: [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/2 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/3 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/4 successfully restarted on span 1 Thanks, Bruce -- _ -- 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 restart of span a concern on PRI?
bruce bruce wrote: [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1 successfully restarted on span 1 It's a normal function: *resetinterval*: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Full transfer details on inbound calls
Hi, This may be no use to you if you are using 1.4 but Call Event Logging (or CEL) that is currently in trunk should provide an easier way to do this. All events associated with a call e.g. Answer, Hangup, Bridge start, Transfer etc. are logged to the usual back-ends. We use postgresql via ODBC. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 13 April 2010 14:50 To: Asterisk Subject: [asterisk-users] Full transfer details on inbound calls Hi We're using asterisk 1.4.17 using RealTime and my boss has decided that we should keep a track of the full history of incoming calls i.e. who and when they were transferred to. The asterisk CDR only holds the initial answering channel for any call and not any further transfers that may have happened. The idea we are toying with is getting the time and the originating channel from the cdr, and then searching the full asterisk logs for the channel identifier string. Obviously we would have to have the verbose output going to a file and make sure that the verbosity in the console is always at least 5. I've done enough testing to see that is is possible i...@trinity:/var/log/asterisk$ grep 'Apr 13' full | grep SIP/xxx.xxx.xxx.xxx-082090e8 | grep answered [Apr 13 13:31:11] VERBOSE[17120] logger.c: -- SIP/811-08214f50 answered SIP/xxx.xxx.xxx.xxx-082090e8 [Apr 13 13:31:31] VERBOSE[17120] logger.c: -- SIP/808-08212f08 answered SIP/xxx.xxx.xxx.xxx-082090e8 The above output shows that the originating channel was answered by sip extension 811 and then by 808 20 seconds later. I am also considering parsing the full log into a mysql database and doing the searching in there. My question is is this a good way to go about what I'm trying to achieve or is there a simpler/less process intensive method that I'm missing. Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Being attacked by an Amazon EC2 ...
Speaking of all these attacks, are there any good web managed security monitor tools for CentOS out there that can be installed on the system so that it can give us a visual of let's multiple failed attempts against SSH or HTTPd? Something nice that is simple and doesn't eat a lot resources and spits out everything on the screen? Thanks, Bruce On Tue, Apr 13, 2010 at 9:51 AM, Fred Posner f...@teamforrest.com wrote: On Apr 13, 2010, at 8:04 AM, Hans Witvliet wrote: On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote: On Tue, 13 Apr 2010, Alyed wrote: Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) I'll agree with you here. Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) So a proper job for ip(6)tables, imho -- +1 for outside of asterisk. I want something that blocks it before it gets to the Asterisk processes. I've posted a little script on Team Forrest for how I'm blocking the traffic (using a quick perl script, iptables, and cron). The script is at http://bit.ly/cDHlLq ---fred http://qxork.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] Is svn.asterisk.org down ?
Hi, Is it me or is svn.asterisk.org down ? 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 restart of span a concern on PRI?
Thanks, I can sleep better now. On Tue, Apr 13, 2010 at 10:02 AM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1 successfully restarted on span 1 It's a normal function: *resetinterval*: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Is svn.asterisk.org down ?
Olivier wrote: Is it me or is svn.asterisk.org http://svn.asterisk.org down ? It is, along with issues.asterisk.org, reviewboard.asterisk.org and some other sites. They should be back up in the next hour. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Being attacked by an Amazon EC2 ...
On Tue, 2010-04-13 at 15:49 +0200, Philipp von Klitzing wrote: Hi! Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) However, I *still* think Asterisk should provide a delayreject option in sip.conf to greatly slow down answering request avanlanches. That will help to address the bandwidth issue if the attacker is configured to wait for a response before starting the next request. Apart from that here are the most important messages: Use strong passwords in sip.conf, and use keys in iax.conf, and avoid usernames that can be guessed too easily (numbers from 100 to and first names). Agreed, best would be to only use ssl-certificates for authentication, but not all parts involved support that, (to put it mildly...) hw -- _ -- 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 svn.asterisk.org down ?
On 13 Apr 2010, at 15:22, Olivier wrote: Is it me or is svn.asterisk.org down ? issues. too -- _ -- 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] dahdi_scan and OctoBRI. Bug or feature ?
On Tue, Apr 13, 2010 at 02:19:26PM +0200, Olivier wrote: Hi, When typing dahdi_scan on an OctoBRI-enabled setup, I've got only 8 replies such as : [1] active=yes alarms=RED ... [8] active=yes ... framing=CCS I would expect 16 replies (one per B-channel). Is this correct ? No. One section per span. For digital spans there's no listing of channels. BTW: do you use qozap from http://junghanns.net/downloads/jnet-dahdi-drivers-1.0.0.tar.gz ? -- 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
Re: [asterisk-users] Is svn.asterisk.org down ?
Olivier escribió: Hi, Is it me or is svn.asterisk.org http://svn.asterisk.org down ? Regards Yep, it's down: mig...@laptop-miguel:~$ ping svn.asterisk.org PING svn.asterisk.org (76.164.171.230) 56(84) bytes of data. From orc2.api-digital.com (63.238.52.42) icmp_seq=1 Destination Host Unreachable -- 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] cat /proc/zaptel/*
On Tue, Apr 13, 2010 at 02:53:51PM +0200, Jaap Winius wrote: Hi all, On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is: ~# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A Zaptel Driver card 1 [TE] AMI/CCS 4 ZTHFC2/0/1 Clear 5 ZTHFC2/0/2 Clear 6 ZTHFC2/0/3 HDLCFCS These are two HFC-S PCI A cards. But, what exactly does all of this mean? In particular: * Span - In telephony, what is the definition of this term? * MASTER - How is this relevant? Only for timing purposes? * Clear - Is this said because only B-channels use ISDN clear codes? * HDLCFCS - Why say this about D-channels? Why not just say HDLC? * (In use) - What does this mean and how is this state determined? * 1 ZTHFC1/0/1 Clear (In use) - What do each of these columns specify? http://docs.tzafrir.org.il/dahdi-linux/#_procfs_interface_proc_dahdi Clear basically means this is an ISDN B-channel. HDLCFCS means this is a D-channel. 'dchan' = 'fcshdlc' . I'll leave aside the issue of 'dchan' vs. hardhdlc because there's actually some chating here. -- 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
Re: [asterisk-users] cat /proc/zaptel/*
On Apr 13, 2010, at 7:53 AM, Jaap Winius wrote: Hi all, On an Asterisk/Zaptel 1.4 system, one way to gather diagnostic info is: ~# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A Zaptel Driver card 1 [TE] AMI/CCS 4 ZTHFC2/0/1 Clear 5 ZTHFC2/0/2 Clear 6 ZTHFC2/0/3 HDLCFCS These are two HFC-S PCI A cards. But, what exactly does all of this mean? In particular: * Span - In telephony, what is the definition of this term? * MASTER - How is this relevant? Only for timing purposes? * Clear - Is this said because only B-channels use ISDN clear codes? * HDLCFCS - Why say this about D-channels? Why not just say HDLC? * (In use) - What does this mean and how is this state determined? * 1 ZTHFC1/0/1 Clear (In use) - What do each of these columns specify? Thanks, Jaap -- _ -- 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 Hi Jaap, Most of these questions are covered in the README in the base of our DAHDI driver directory. The most up-to-date readme is maintained by Tzafrir at http://docs.tzafrir.org.il/dahdi-linux/README.html 1) A span is a physical plug. It can either be a single channel analog port or a T1/E1 port with lots of channels. 2) MASTER is zaptel/dahdi's current source of timing for the entire system. This is what Asterisk will use to time meetme conferences. 3) Clear means that there is no signaling or anything on that channel, so the data is provided and can be read as is 4) It's the specific type of hdlc your hardware is using? You could source dive for more info 5) This means that asterisk is currently using the channel, probably determined by a file lock --- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer -- _ -- 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] SIP equivalent of zap c option
At the moment, we have a feature where if someone's sip extension is called, we also make another call to their mobile. We use the c option in the zap dialstring so that the user has to press # after answering to confirm the call (this prevents things like the answermachine grabbing the call if the mobile is switched off). We are now looking to move towards a sip provider to take all of our ISDN calls, so instead of using zap / isdn to call the mobile, we will be routing the call over a SIP trunk. Is there any feature of SIP that we can use in order to duplicate this functionality (i.e. have to press # to confirm the call) Thanks Julian -- _ -- 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] Being attacked by an Amazon EC2 ...
- Original Message - Speaking of all these attacks, are there any good web managed security monitor tools for CentOS out there that can be installed on the system so that it can give us a visual of let's multiple failed attempts against SSH or HTTPd? Something nice that is simple and doesn't eat a lot resources and spits out everything on the screen? Thanks, Bruce How about http://www.ossec.net which you could later integrate with http://www.splunk.com/. -- Thanks - Phil -- _ -- 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] Being attacked by an Amazon EC2 ...
Cool. I am just looking over splunk. Isn't that enough by it's own? or is OSSEC needed to give it raw data? I think these two will take quite some time to understand. Anything simpler out there as well? Thanks, Bruce On Tue, Apr 13, 2010 at 10:42 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: - Original Message - Speaking of all these attacks, are there any good web managed security monitor tools for CentOS out there that can be installed on the system so that it can give us a visual of let's multiple failed attempts against SSH or HTTPd? Something nice that is simple and doesn't eat a lot resources and spits out everything on the screen? Thanks, Bruce How about http://www.ossec.net which you could later integrate with http://www.splunk.com/. -- Thanks - Phil -- _ -- 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] Being attacked by an Amazon EC2 ...
- Original Message - Cool. I am just looking over splunk. Isn't that enough by it's own? or is OSSEC needed to give it raw data? I think these two will take quite some time to understand. Anything simpler out there as well? Thanks, Bruce On Tue, Apr 13, 2010 at 10:42 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: - Original Message - Speaking of all these attacks, are there any good web managed security monitor tools for CentOS out there that can be installed on the system so that it can give us a visual of let's multiple failed attempts against SSH or HTTPd? Something nice that is simple and doesn't eat a lot resources and spits out everything on the screen? Thanks, Bruce How about http://www.ossec.net which you could later integrate with http://www.splunk.com/ . OSSEC has a number of Asterisk rules already built it; including picking up failed SIP registrations. It also has the feature called Active Response which when a user defined threshold of failed events happen it is able to automatically add a IPtables/PF drop rule for the source IP. -- Thanks, Phil -- _ -- 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 equivalent of zap c option
Hi,
On Tue, Apr 13, 2010 at 03:37:37PM +0100, Julian Lyndon-Smith wrote:
At the moment, we have a feature where if someone's sip extension is
called, we also make another call to their mobile. We use the c
option in the zap dialstring so that the user has to press # after
answering to confirm the call (this prevents things like the
answermachine grabbing the call if the mobile is switched off).
We are now looking to move towards a sip provider to take all of our
ISDN calls, so instead of using zap / isdn to call the mobile, we will
be routing the call over a SIP trunk. Is there any feature of SIP that
we can use in order to duplicate this functionality (i.e. have to
press # to confirm the call)
You could do something like this:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
It works well for me. If I answer then hit 1 on my cellphone I
get the call, otherwise it goes to Asterisk VM, and never to
the cellphone VM.
[whatever]
exten = s,n,Dial(${EXT}${CCME}local/1...@internals,20,rt)
[internals]
exten = 101,1,Dial(${MARKCELL},30,tgM(screen))
exten = 101,n,Goto(main-menu,s,1) ; if not answered and accepted
[macro-screen]
exten = s,1,Wait(0.5)
exten = s,n,Read(ACCEPT,followme/options,1,,1,20)
exten = s,n,GotoIf($[${ACCEPT} = 1]?yes:no)
exten = s,n(yes),Background(connecting)
exten = s,n,Goto(end)
exten = s,n(no),Set(MACRO_RESULT=CONTINUE)
exten = s,n(end),NoOp
Mark
--
Mark G. Thomas (m...@misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112
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Re: [asterisk-users] Full transfer details on inbound calls
Hi! We're using asterisk 1.4.17 using RealTime and my boss has decided that we should keep a track of the full history of incoming calls i.e. who and when they were transferred to. The asterisk CDR only holds the initial answering channel for any call and not any further transfers that may have happened. The idea we are toying with is getting the time and the originating channel from the cdr Depending on the type of phones you use there is also another way to look at it: SNOM phones for example have event triggers for certain actions that can call a URL, this includes events like off-hook, transfer completed and the like. Taking it even further: You could try trace all SIP signaling to/from your Asterisk, or even on your entire LAN, and then try to analyse that. I believe there's a number of SIP tools available for that. Philipp -- _ -- 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] Problem with Callfiles
Hi! I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt. I put here my callfile and that I get when asterisk begins to do the call If anybody has idea, pls. Tell me TIA ;;CallFile- Channel: Zap/g1/8093908270 Callerid: 8093908270 MaxRetries: 2 RetryTime: 300 WaitTime: 45 Context: 1call Extension: s Priority: 1 ;;EXTENSION:: [1call] exten = s,1,Playback(vm-intro) exten = s,2,Playback(vm-goodbye) exten = s,3,Hangup I am getting this when I put the 1.call to outgoing directory. The call never started == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.30 currently running on ivr-server (pid = 1873) Verbosity is at least 5 Channel Zap/8-1 was answered. -- Executing [...@1call:1] Playback(Zap/8-1, vm-intro) in new stack -- Zap/8-1 Playing 'vm-intro' (language 'en') -- Executing [...@1call:2] Playback(Zap/8-1, vm-goodbye) in new stack -- Zap/8-1 Playing 'vm-goodbye' (language 'en') -- Executing [...@1call:3] Hangup(Zap/8-1, ) in new stack == Spawn extension (1call, s, 3) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' [Apr 13 00:54:03] NOTICE[2493]: pbx_spool.c:370 attempt_thread: Call completed to Zap/g1/8093908270 I tested the channel doing a call to this and I get this, the call worked -- Starting simple switch on 'Zap/8-1' [Apr 13 00:58:27] NOTICE[2496]: chan_dahdi.c:6869 ss_thread: Got event 18 (Ring Begin)... [Apr 13 00:58:28] NOTICE[2496]: chan_dahdi.c:6869 ss_thread: Got event 2 (Ring/Answered)... [Apr 13 00:58:32] NOTICE[2496]: chan_dahdi.c:6869 ss_thread: Got event 18 (Ring Begin)... -- Executing [...@from-pstn:1] Answer(Zap/8-1, ) in new stack -- Executing [...@from-pstn:2] Playback(Zap/8-1, vm-intro) in new stack -- Zap/8-1 Playing 'vm-intro' (language 'en') -- Executing [...@from-pstn:3] Hangup(Zap/8-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Merge master.csv files
Hi there, Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files? Thanks in advanced, Ricardo Coelho -- _ -- 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] Using chan_lcr (and mISDN v2) ?
Hi, A new http://misdn.org/index.php/Howto_for_Debian doc has been published Along with http://www.linux-call-router.de/howto.html, it describes a way to install Asterisk along mISDN V2. Has someone experienced with it ? Thoughts ? Could it be a reliable path for alternate ISDN devices like AVM boards ? 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
[asterisk-users] Merge .csv files
Hi there, Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files? Thanks in advanced, Ricardo Coelho -- _ -- 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] Using chan_lcr (and mISDN v2) ?
Hi! to install Asterisk along mISDN V2. Has someone experienced with it ? Thoughts ? Could it be a reliable path for alternate ISDN devices like AVM boards ? If you take a look at the misdn mailing list you will see that the future of mISDN v2 is quite uncertain - at least it was when I looked at it a couple of weeks ago. Technically it is a much more sound solution than mISDN v1, but personally I would really not want to use it until also a chan_misdn v2 becomes available. Currently you need to run LCR (linux call router) in parallel to Asterisk in order to make use of mISDN v2. Philipp -- _ -- 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] Being attacked by an Amazon EC2 ...
On Tue, Apr 13, 2010 at 04:32:58PM +0200, Hans Witvliet wrote: On Tue, 2010-04-13 at 15:49 +0200, Philipp von Klitzing wrote: Hi! Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) However, I *still* think Asterisk should provide a delayreject option in sip.conf to greatly slow down answering request avanlanches. That will help to address the bandwidth issue if the attacker is configured to wait for a response before starting the next request. Apart from that here are the most important messages: Use strong passwords in sip.conf, and use keys in iax.conf, and avoid usernames that can be guessed too easily (numbers from 100 to and first names). Agreed, best would be to only use ssl-certificates for authentication, but not all parts involved support that, (to put it mildly...) Secure authentication won't solve the problem of attackers flodding your pipe. Especially not if you have ADSL or similar connection. -- 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] Time variables in system application
Hi Guys, i have a weird thing here: when using time variables (%F %T) in a shell script, out of dial plan (particularly system() app); it displays the right time (same as output of date), but when same variables are used in system() application it displays a wrong time/date (ahead of 6 hours). I am using a centos 5.3, can anyone help me fix this? -- Abdullah -- _ -- 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] iptables miss up phone calls if not used properly
Hi Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 2, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech other (but not all the time which weird), so iptables can miss up performance if not set correctly (even if it's working, stuff like this can happen). so if any body have some lines of iptables that secure server and don't cause performence trouble to phone calls please share with me (i am using Centos 5.3 asterisk 1.4.24). Thanks! -- Abdullah -- _ -- 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] Being attacked by an Amazon EC2 ...
Hmmm. It would seem that it would be to Amazon's advantage to jump on this problem, because the accounts that are performing this activity are most likely purchased with stolen identities, and sooner or later the charges are going to get reversed. Either the credit card companies are going to absorb the cost, or the merchants (like Amazon) at the other end. And, after listening to merchants grumble about it, I'd assume that in the end, Amazon is going to get stiffed for the bill. On someone else's credit card, I'd imaging they have almost infinite resources; Bandwidth to burn, the best and most powerful hosts. So what if they rack up thousands of dollars? They are probably organized crime units in Romania or whatever. murf On Tue, Apr 13, 2010 at 11:21 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Apr 13, 2010 at 04:32:58PM +0200, Hans Witvliet wrote: On Tue, 2010-04-13 at 15:49 +0200, Philipp von Klitzing wrote: Hi! Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) However, I *still* think Asterisk should provide a delayreject option in sip.conf to greatly slow down answering request avanlanches. That will help to address the bandwidth issue if the attacker is configured to wait for a response before starting the next request. Apart from that here are the most important messages: Use strong passwords in sip.conf, and use keys in iax.conf, and avoid usernames that can be guessed too easily (numbers from 100 to and first names). Agreed, best would be to only use ssl-certificates for authentication, but not all parts involved support that, (to put it mildly...) Secure authentication won't solve the problem of attackers flodding your pipe. Especially not if you have ADSL or similar connection. -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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] Time variables in system application
You are apparently in U.S. Central Time zone.Asterisk uses the hardware clock and system() uses the system clock, so these are probably out of sync. Try doing Date and Hwclock From a command prompt. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Tuesday, April 13, 2010 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Time variables in system application Hi Guys, i have a weird thing here: when using time variables (%F %T) in a shell script, out of dial plan (particularly system() app); it displays the right time (same as output of date), but when same variables are used in system() application it displays a wrong time/date (ahead of 6 hours). I am using a centos 5.3, can anyone help me fix this? -- Abdullah -- _ -- 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] Time variables in system application
You are apparently in U.S. Central Time zone.Asterisk uses the hardware clock and system() uses the system clock, so these are probably out of sync. Try doing Date and Hwclock From a command prompt. thanks, here is the output of the two clocks you mentioned they dispaly same info (slight diff on in 24 and other 12 format)!! if any body know what's the issue, i will be grateful! [r...@pbx1 bin]# hwclock Tue 13 Apr 2010 02:40:16 PM EDT -0.000607 seconds [r...@pbx1 bin]# date Tue Apr 13 14:41:11 EDT 2010 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati *Sent:* Tuesday, April 13, 2010 1:08 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Time variables in system application Hi Guys, i have a weird thing here: when using time variables (%F %T) in a shell script, out of dial plan (particularly system() app); it displays the right time (same as output of date), but when same variables are used in system() application it displays a wrong time/date (ahead of 6 hours). I am using a centos 5.3, can anyone help me fix this? -- Abdullah -- _ -- 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 -- Abdullah -- _ -- 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] Do AMI Events have timestamps?
I have been monitoring AMI events and realized that they don't have timestamps. Is that standard behaviour, or is there some way to get them to include timestamps? I am on 1.4. Is it available on 1.6? -- _ -- 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] iptables miss up phone calls if not used properly
On Tue, 13 Apr 2010, khalid touati wrote: Hi Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 2, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech other (but not all the time which weird), so iptables can miss up performance if not set correctly (even if it's working, stuff like this can happen). so if any body have some lines of iptables that secure server and don't cause performence trouble to phone calls please share with me (i am using Centos 5.3 asterisk 1.4.24). You've probably blocks too much and it's stopping DNS working properly. Gordon -- _ -- 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] Time variables in system application
On Tuesday 13 April 2010 13:27:30 Danny Nicholas wrote: You are apparently in U.S. Central Time zone.Asterisk uses the hardware clock What makes you think Asterisk uses the hardware clock? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 DIES with no trace. PLEASE
Hello Zeeshan/Asterisk-users We are having a little problem in our Asterisk pbx using our A102DE, just like Zeeshan told us about problems with zap, even if a zap channel is in use the Hookstat is always onhook, never changes to offhook If the line is in use or not, the behavior of the Hookstate is always onhook, is this a problem? what should we do? MyPbx*CLI zap show channel 31 Channel: 31I File Descriptor: 44 Span: 1 Extension: I Dialing: no Context: mde-g0 Caller ID: 2432690033 Calling TON: 33 Caller ID name: Destroy: 0LI InAlarm: 0 Signalling Type: ISDN PRI Radio: 0*CLI Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON PRI Flags: I PRI Logical Span: Implicit Hookstate (FXS only): Onhook Thanks in advance! Message: 1 Date: Thu, 18 Mar 2010 11:20:38 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Do you properly hang up the calls. Does 'zap show channel channel number' shows that the channel is 'on hook' after its hang up? On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: 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 -- _ -- 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] Do AMI Events have timestamps?
They actually do have a timestamp, in a manner of speaking. The uniqueid field is a pseudo-unixtime stamp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church Sent: Tuesday, April 13, 2010 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Do AMI Events have timestamps? I have been monitoring AMI events and realized that they don't have timestamps. Is that standard behaviour, or is there some way to get them to include timestamps? I am on 1.4. Is it available on 1.6? -- _ -- 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] Time variables in system application
Just what I thought - guess that's the X'th time I wuz wrong today. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, April 13, 2010 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Time variables in system application On Tuesday 13 April 2010 13:27:30 Danny Nicholas wrote: You are apparently in U.S. Central Time zone.Asterisk uses the hardware clock What makes you think Asterisk uses the hardware clock? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Merge .csv files
On Tue, Apr 13, 2010 at 12:56 PM, Ricardo Coelho ricardo.tch...@gmail.com wrote: Hi there, Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files? Yes. download asterisk-extras compile cdr_mysql setup a shared database, point both systems at that shared database. If you're going to do anything even moderately advanced with processing your csv files, you'll be glad you went ahead and put this stuff into a database. Or you can skip the cdr_mysql, but manually dump two Master.csv files into a database to play with, if you don't mind your database not continuing to update with new info. -- _ -- 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] Possible AGI bug?
Hello all, I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part of an audio file, sometimes gets stuck as if waiting for something to happen. I've run the script with verbose=3 and agi debug on, but couldn't detect anything abnormal. How does one even start to debug a problem like this? I find it really bizarre that the system behaves differently even if called in the same way... Any suggestions? Thanks in advance, Leo -- _ -- 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] iptables miss up phone calls if not used properly
DNS!! i believe it has to do with call setup and rtp protocol cause all devices shows as sip peers at the call time, but not 100% sure. any iptables plz :) ! 2010/4/13 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Tue, 13 Apr 2010, khalid touati wrote: Hi Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 2, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech other (but not all the time which weird), so iptables can miss up performance if not set correctly (even if it's working, stuff like this can happen). so if any body have some lines of iptables that secure server and don't cause performence trouble to phone calls please share with me (i am using Centos 5.3 asterisk 1.4.24). You've probably blocks too much and it's stopping DNS working properly. Gordon -- _ -- 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 -- Abdullah -- _ -- 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] Possible AGI bug?
Is the Originate() call using the same context as the manual Dial-In? Could be as simple as one Answering and the other not (or not always). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leo Burd Sent: Tuesday, April 13, 2010 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Possible AGI bug? Hello all, I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part of an audio file, sometimes gets stuck as if waiting for something to happen. I've run the script with verbose=3 and agi debug on, but couldn't detect anything abnormal. How does one even start to debug a problem like this? I find it really bizarre that the system behaves differently even if called in the same way... Any suggestions? Thanks in advance, Leo -- _ -- 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] Do AMI Events have timestamps?
On Tue, 2010-04-13 at 13:59 -0500, Danny Nicholas wrote: They actually do have a timestamp, in a manner of speaking. The uniqueid field is a pseudo-unixtime stamp. While correct, it's a timestamp of when the call *started*, not when the event happened. -- Jared Smith Digium, Inc. -- _ -- 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] Time variables in system application
At least on this forum, bad help usually leads to good help??? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Tuesday, April 13, 2010 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Time variables in system application i believe not only today :D, but thank u anyway for the spirit of helping people!! 2010/4/13 Danny Nicholas da...@debsinc.com Just what I thought - guess that's the X'th time I wuz wrong today. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, April 13, 2010 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Time variables in system application On Tuesday 13 April 2010 13:27:30 Danny Nicholas wrote: You are apparently in U.S. Central Time zone.Asterisk uses the hardware clock What makes you think Asterisk uses the hardware clock? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- Abdullah -- _ -- 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] Time variables in system application
i believe not only today :D, but thank u anyway for the spirit of helping people!! 2010/4/13 Danny Nicholas da...@debsinc.com Just what I thought - guess that's the X'th time I wuz wrong today. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, April 13, 2010 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Time variables in system application On Tuesday 13 April 2010 13:27:30 Danny Nicholas wrote: You are apparently in U.S. Central Time zone.Asterisk uses the hardware clock What makes you think Asterisk uses the hardware clock? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- Abdullah -- _ -- 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] DAHDI-Linux and DAHDI-Tools 2.3.0 Released
The Asterisk Development Team is pleased to announce the release of DAHDI-Linux and DAHDI-Tools version 2.3.0. DAHDI-Linux 2.3.0, DAHDI-Tools 2.3.0, and DAHDI-Linux-Complete are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete In addition to several bug fixes, the most significant changes from the 2.2.0 release are: General DAHDI Changes: * Static /dev/dahdi files are not generated at install time since udev is used on all the supported distributions. build_tools/make_static_devs is available for those users who still need the static device files. * UDEV_DIR can be set during build in order to override where the build will place the udev rules. * dahdi_dummy is no longer built by default. DAHDI will automatically use a kernel timer to provide timing if there isn't a physical span which is providing timing. * Added support for 16 kbps software hdlc. * Added support for software configurable BRI TE/NT mode and termination resistance. * Support for additional error counts added to dahdi_spaninfo. New counters include framing errors, coding violations, bit errors, and errored seconds. New Drivers: * dahdi_dynamic_ethmf included to support TDMoE Multi-Frame Devices from Redfone Communications. [http://www.thrallingpenguin.com/articles/tdmoe-mf.htm] Updated Drivers: * wctdm24xxp: Support for Digium Hx8 hybrid digital/analog cards. [http://www.digium.com/en/products/hybrid/] * wcte12xp, wct4xxp: Added support for more fine grained maintenance modes. Among other things, it is possible to now place cards supported by these drivers into local loopback without the use of an external loopback plug. See dahdi_maint in dahdi-tools for more information. * wct4xxp: Added support for Fifth Generation firmware which allows dual and quad span cards to function on systems which are unable to service the interrupt every millisecond in addition to support for revision 3.1 of the framer. * wcb4xxp: Swyx 4xS0 SX2 QuadBri, HFC-4S Eval board, and several additional Junghanns cards added to the device table. * wct4xxp, wcte12xp: Added losalarmdebounce, aisalarmdebounce, and yelalarmdebounce module parameters in order to configure alarm debounce times (specified in ms). * wcte12xp: Added 'max_latency' module parameter. VPM initialization moved into start span and VPM is polled and reset if necessary. This is to accommodate systems that experience interrupt latencies 128ms. * xpp: 'offhook' also applies to the PRI modules - no PCM passed if no call. Udev rules updated for newer kernels / udev. Changes to dahdi-tools: * dahdi_maint: A new tool which allows the maintenance mode of spans that support the maintenance mode interface to be configured. This includes setting various loopback modes and error injection. * dahdi_tool: loop button was removed from the span page. This will soon be handled through the dahdi_maint utility. * fxstest: now supports generating DTMF CID spills for testing. * dahdi-perl: /proc/bus/usb no longer required for dahdi_hardware and such. Can use a dump generated by build_tools/dump_sys_state. Support loading firmware from 1163 (minimal firmware) devices. For a full list of changes in these releases, please see the ChangeLogs at http://svn.asterisk.org/svn/dahdi/linux/tags/2.3.0/ChangeLog and http://svn.asterisk.org/svn/dahdi/tools/tags/2.3.0/ChangeLog Issues found in these releases can be reported at http://issues.asterisk.org Thank you for your continued support of Asterisk! -- _ -- 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] Time variables in system application
On Tuesday 13 April 2010 14:00:36 Danny Nicholas wrote: Just what I thought - guess that's the X'th time I wuz wrong today. The only difference between what I think you're calling the system time (output of date) and Asterisk is that Asterisk uses a different (internal) library to convert the epoch-based time into a broken-out date. Both are using exactly the same value internally, however. Hardware clock is generally how system time is set initially at boot, though with NTP servers and system skew, it's possible for the two values to drift apart over time. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Do AMI Events have timestamps?
Would making timestamp=yes in manager.conf have any effect on this behavior? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith Sent: Tuesday, April 13, 2010 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Do AMI Events have timestamps? On Tue, 2010-04-13 at 13:59 -0500, Danny Nicholas wrote: They actually do have a timestamp, in a manner of speaking. The uniqueid field is a pseudo-unixtime stamp. While correct, it's a timestamp of when the call *started*, not when the event happened. -- Jared Smith Digium, Inc. -- _ -- 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] Time variables in system application
My derailed train of thought came from OP's mention of Centos 5.3 - I have to do a hwclock -s on my 5.3 box at least daily to keep a reasonable time. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, April 13, 2010 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Time variables in system application On Tuesday 13 April 2010 14:00:36 Danny Nicholas wrote: Just what I thought - guess that's the X'th time I wuz wrong today. The only difference between what I think you're calling the system time (output of date) and Asterisk is that Asterisk uses a different (internal) library to convert the epoch-based time into a broken-out date. Both are using exactly the same value internally, however. Hardware clock is generally how system time is set initially at boot, though with NTP servers and system skew, it's possible for the two values to drift apart over time. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Problems with Fax over TDM410P
What do you mean with problems on my configuration? This is a FXO port on zapata: signalling=fxs_ks group=0 channel = 1 Not a FXS...can you explain to me what were you trying to say? Message: 4 Date: Mon, 12 Apr 2010 13:14:49 -0400 From: David Backeberg dbackeb...@gmail.com Subject: Re: [asterisk-users] Problems with Fax over TDM410P To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: p2l3de056a31004121014jc8037ab7sb9f84cc9d...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 On Sun, Apr 11, 2010 at 5:00 PM, Danny Dias ing.diasda...@gmail.com wrote: This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed in this case! ; TDM410P signalling=fxs_ks group=0 channel = 1 Signalling=fxs_ks group=0 channel = 2 signalling=fxs_ks group=0 channel = 3 signalling=fxo_ks group=1 channel = 4 What should we do in order to make it work ok? we really need to put this If you really have three FXO, and one FXS, there's part of your problem. You have your zapata configured as three FXS and one FXO. I would suspect that would be a good enough reason to crash your card or whatever. -- _ -- 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] Being attacked by an Amazon EC2 ...
On Tue, Apr 13, 2010 at 8:25 PM, Steve Murphy m...@parsetree.com wrote: Hmmm. It would seem that it would be to Amazon's advantage to jump on this problem, I am pushing for this, please everyone who is suffering from this problem, submit it or write to complain to Amazon and post the message publicly wherever you can in a civilized, even lucid message to them. If you do it they will take notice. They need to see this as a problem in their space and take reasonable steps to either make it harder to abuse their service and/or easier to report the abuse, which they must then act upon. The thread here is an interesting discussion, but it can't compare to actual action they might take if your complaints reach them. They will need to act, but only if you force them to take notice. I believe Amazon has a chance to distinguish themselves from ISP who allow spammers to do mass mailings without any real challenge. They will act if you continue putting the message out there. /r -- _ -- 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] SIP registration failure stops all SIP activity
I have a problem that when one of my SIP providers has a problem the rest of my SIP extensions and trunks stop working until either the SIP provider fixes the problem or Asterisk stops trying to register to that provider. Why does this happen? A single provider having problems should not grind everything else to a halt! At this moment I either have to comment the register lines for that provider or wait until the registration times out (I have 10 attempts and 60 second delay in sip.conf). During that time all sip phones have no service and other trunk providers (SIP) are all UNREACHABLE. Is there something I can change in my sip.conf to prevent this problem? -- 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] Merge master.csv files
On Tue, 13 Apr 2010, Ricardo Coelho wrote: Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files? It is not worth the effort to find out if Asterisk closes the file -- it's a major league bad idea. At some point, you will discover a race condition at the application, network daemon, file system, or OS level. Dumping the CDRs into a database is a much better idea. The benefits go way beyond concurrency. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Possible AGI bug?
On Tue, 13 Apr 2010, Leo Burd wrote: I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part of an audio file, sometimes gets stuck as if waiting for something to happen. Whose AGI library did you use? Violating the protocol can introduce difficult to debug bugs. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] iptables miss up phone calls if not used properly
On Tue, Apr 13, 2010 at 11:17 AM, khalid touati khalidtou...@gmail.com wrote: Hi Guys, i wanted to share this with u and ask for little help at the same time: i used iptables to secure my server, so i wnet ahead and blocked avery thing except a couple of domain protocols and UDP ports of SIP, IAX2 and that range 15000 to 2, tested it and OK. when in production, the calls were taking a huge time 7s to be established and somtimes after call setup people cannot hear ech other (but not all the time which weird), so iptables can miss up performance if not set correctly (even if it's working, stuff like this can happen). so if any body have some lines of iptables that secure server and don't cause performence trouble to phone calls please share with me (i am using Centos 5.3 asterisk 1.4.24). You don't need to open up all of the UDP ports like that if you enable connection tracking for sip. Of course you don't say how many ongoing sessions you are using, but I haven't had any issues with connection tracking for SIP. All of this is based on INBOUND connections to the server, but make sure you are allowing OUTBOUND connections too. Here are some changes for an example that is NOT complete and you can use AT YOUR OWN RISK. Make sure you have something like this in the following files. Notice that this does not restrict who can talk to your server either, and only covers IAX/SIP. This is based on CentOS 5.4. /etc/sysconfig/iptables: # Anything we already know about -A Fwall-IN -m state --state ESTABLISHED,RELATED -j ACCEPT # IAX -A Fwall-IN -m state --state NEW -m udp -p udp --dport 4569 -j ACCEPT # SIP -A Fwall-IN -m state --state NEW -m udp -p udp --dport 5060 -j ACCEPT -A Fwall-IN -m state --state NEW -m tcp -p tcp --dport 5060 -j ACCEPT /etc/sysconfig/iptables-config: IPTABLES_MODULES=ip_conntrack_sip If you need more specifics, you will have to post your iptables configuration for some more advise. -Jonathan -- _ -- 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] Problems with Fax over TDM410P
check the IRQ and make sure the TDM410P has it owns IRQ. From: Danny Dias ing.diasda...@gmail.com To: asterisk-users@lists.digium.com Sent: Fri, April 9, 2010 4:52:05 PM Subject: [asterisk-users] Problems with Fax over TDM410P Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed in this case! take a look in our zapata: [channels] language=es ;context=default rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no busydetect=yes immediate=no ;busycount=4 ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ; TDM410P context = mde-g1 immediate=no signalling=fxs_ks group=0 channel = 1 context = mde-g1 immediate=yes Signalling=fxs_ks group=0 channel = 2 context = mde-g1 immediate=yes signalling=fxs_ks group=0 channel = 3 context=inside faxdetect=incoming immediate=no signalling=fxo_ks group=1 channel = 4 What should we do in order to make it work ok? we really need to put this working, i've heard that asterisk does not work very well with fax, but at least it should try to dend it, not to get frozen :S Thanks in advance for all your help! 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] Being attacked by an Amazon EC2 ...
On Apr 13, 2010, at 4:22 PM, Randy R wrote: On Tue, Apr 13, 2010 at 8:25 PM, Steve Murphy m...@parsetree.com wrote: Hmmm. It would seem that it would be to Amazon's advantage to jump on this problem, I am pushing for this, please everyone who is suffering from this problem, submit it or write to complain to Amazon and post the message publicly wherever you can in a civilized, even lucid message to them. If you do it they will take notice. They need to see this as a problem in their space and take reasonable steps to either make it harder to abuse their service and/or easier to report the abuse, which they must then act upon. The thread here is an interesting discussion, but it can't compare to actual action they might take if your complaints reach them. They will need to act, but only if you force them to take notice. I believe Amazon has a chance to distinguish themselves from ISP who allow spammers to do mass mailings without any real challenge. They will act if you continue putting the message out there. /r The only person I've gotten to respond to me is Kay Kinton from Amazon's Public Relations. Although she responded, she will not take a phone call or discuss the issue over the phone. She gave me two statements so far, which I will be posting on VoIPTechChat.com (one's there already). Statement 1: Hello Fred and thank you for contacting us. Over the weekend, we received a report of a suspicious account and began an investigation. Our normal process is to connect the two involved parties to give them an opportunity to talk in case the abuse is not malicious but is simply heavy traffic from a legitimate customer. If that is not successful, we then move to isolate the traffic from the abusing party. Normally this process works quite well for situations our customers have encountered, however this incident has highlighted the need for an escalation process to address potentially malicious attacks more quickly. Additionally, we are working on quickly putting better protections and processes in place to better guard against unwanted SIP traffic. We take the security of our customers and our quality of service very seriously, and will continue to work to improve our processes and services for customers. /end statement 1 This was of course was while attacks were continuing so I asked for a discussion and sent her several questions when she told me what else can I tell you. Today I received statement 2: Hello Fred. We believe that we've identified and shut down the illegal activity and are closing the loop with customers. We'd certainly be interested in hearing of the cases you refer to below so we can follow up. /end statement 2. So.. since she's interested... please let her know how they did not respond to your complaints, the attacks, and well, any of the concerns you have to which she should follow up: Kay Kinton kin...@amazon.com Public Relations Manager Amazon Web Services Phone: 206-266-8387 ---fred http://qxork.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] Possible AGI bug?
- I've just learned that my system now seems to work perfectly fine if I call AMI Originate with $channel = 'Local/%num...@vd-dial_out'; instead of $channel = 'Local/%num...@vd-dial_out/n'; // Note the extra /n at the end I thought it was important to use '/n' to avoid weird behavior (check http://www.voip-info.org/tiki-index.php?page=Asterisk%20local%20channels), but now I'm confused. Any ideas about what is going on? Thanks so much, Leo Steve Edwards wrote: On Tue, 13 Apr 2010, Leo Burd wrote: I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part of an audio file, sometimes gets stuck as if waiting for something to happen. Whose AGI library did you use? Violating the protocol can introduce difficult to debug bugs. -- _ -- 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] What's are the possible return values of AMI Originate when Async is set to 0?
Hello all, What are the possible values returned by AMI Originate when it's called with Async set to 0? Is there any way to find out whether the dialed channel was busy, invalid, etc. without requiring Async to be 1? Thanks in advance, Leo -- _ -- 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 DIES with no trace. PLEASE
Sorry, the last message was incomplete. So with AMI encoding the Rhino card wouldn't work reliably, on which they were able to send us new zaptel drivers patched for our use. That fixed the issue on our end. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-13 5:25 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hi Danny, Actually the issue I faced was the opposite, i.e. the channels would stay offhook even after the hangup. Now I can't remember all the details but that setup had a lot of problems, primarily because it was a very customized system, and the Rhino T1 card was not able to correctly work with E1 when it was used with AMI signalling. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-13 3:03 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello Zeeshan/Asterisk... -- _ -- 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 DIES with no trace. PLEASE
Hi Danny, Actually the issue I faced was the opposite, i.e. the channels would stay offhook even after the hangup. Now I can't remember all the details but that setup had a lot of problems, primarily because it was a very customized system, and the Rhino T1 card was not able to correctly work with E1 when it was used with AMI signalling. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-13 3:03 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello Zeeshan/Asterisk-users We are having a little problem in our Asterisk pbx using our A102DE, just like Zeeshan told us about problems with zap, even if a zap channel is in use the Hookstat is always onhook, never changes to offhook If the line is in use or not, the behavior of the Hookstate is always onhook, is this a problem? what should we do? MyPbx*CLI zap show channel 31 Channel: 31I File Descriptor: 44 Span: 1 Extension: I Dialing: no Context: mde-g0 Caller ID: 2432690033 Calling TON: 33 Caller ID name: Destroy: 0LI InAlarm: 0 Signalling Type: ISDN PRI Radio: 0*CLI Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON PRI Flags: I PRI Logical Span: Implicit Hookstate (FXS only): Onhook Thanks in advance! Message: 1 Date: Thu, 18 Mar 2010 11:20:38 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Do you properly hang up the calls. Does 'zap show channel channel number' shows that the channel is 'on hook' after its hang up? On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: 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 -- _ -- 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] DAHDI-Linux and DAHDI-Tools 2.3.0 Released
On Tue, 13 Apr 2010, Asterisk Development Team wrote: * Static /dev/dahdi files are not generated at install time since udev is used on all the supported distributions. build_tools/make_static_devs is available for those users who still need the static device files. Please do not ever remove the static_devs script - I do not use udev and never will in my embedded systems. There's simply no need for it when your hardware never changes. Gordon -- _ -- 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] Interesting One Way Audio
I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user dials a number that autoanswers line 1-800-555-1212 the Mitel user will hear audio for 1/2 a second then it is dropped. I turned of iptables and it acts the same way. Anyone have any ideas? -- -Thermal -- _ -- 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] 1.6.0 verses 1.6.2
Why do versions 1.6.2 and 1.6.1 use much more CPU resources that 1.6.0? I can get 400+ SIP/G.711 calls running on this dual core box with 1.6.0 but the cpu maxes out and core dumps at approx. 180 calls when version 1.6.1/2 is running. John -- _ -- 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] tones detection
How about a generic beep detector? One that detects beeps at various frequencies not fixed frequencies that would listen to the RTP audio stream and send out a manager event when a detection occurs? John -Original Message- Hi Jerry, On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis ge...@pagestation.com wrote: I am looking for something in asterisk that will let me record a wav file in asterisk (which I know how to do) then some other command (external or dialplan) that would read the wave file and tell me if a certain tone or frequency is present. Is this in asterisk already - any way to do it? Thanks You might want to look into the PipeWave tools: http://www.cardiff.ac.uk/psych/home2/CullingJ/pipewave.html The tools can generate a FFT (fast-fourier transform) of a wav file which converts the data into the frequency domain, which should allow you to tell if a certain frequency is present. -- James Jerry -- -- _ -- 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] Problems with Fax over TDM410P
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote: What do you mean with problems on my configuration? This is a FXO port on zapata: signalling=fxs_ks group=0 channel = 1 Not a FXS...can you explain to me what were you trying to say? http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#SignallingType Yep. If you say that's an fxo port, that's a disagreement between what you told me and what you told the DAHDI layer. You told DAHDI it's fxs. Try changing the config to say fxo and tell us what happens. -- _ -- 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] Problems with Fax over TDM410P
On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg dbackeb...@gmail.com wrote: On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote: What do you mean with problems on my configuration? This is a FXO port on zapata: signalling=fxs_ks group=0 channel = 1 Not a FXS...can you explain to me what were you trying to say? http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#SignallingType Yep. If you say that's an fxo port, that's a disagreement between what you told me and what you told the DAHDI layer. You told DAHDI it's fxs. Try changing the config to say fxo and tell us what happens. Of course, after re-reading what I just wrote, I think I have it backwards. My advice to flip the config and see what happens still applies. Does a regular call work fine? -- _ -- 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] Problems with Fax over TDM410P
On Tue, Apr 13, 2010 at 06:59:01PM -0400, David Backeberg wrote: On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg dbackeb...@gmail.com wrote: On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote: What do you mean with problems on my configuration? ?This is a FXO port on zapata: signalling=fxs_ks group=0 channel = 1 Not a FXS...can you explain to me what were you trying to say? http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#SignallingType Yep. If you say that's an fxo port, that's a disagreement between what you told me and what you told the DAHDI layer. You told DAHDI it's fxs. Try changing the config to say fxo and tell us what happens. Of course, after re-reading what I just wrote, I think I have it backwards. You do. FXO ports want fxs signalling, and vice-versa. -- _ -- 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
