Re: [asterisk-users] Recommended VOIP Monitoring Tools
Smokeping with sip probe is quite nice Sent from BETA iOS6 On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On 12-07-13 08:37 AM, Mike wrote: On 12-07-13 06:00 AM, Elliot Murdock wrote: Hello, Which tools are recommendable for monitoring VOIP, bandwidth, server alarms, etc.? Nagios (http://www.nagios.org/) can be configured to monitor pretty much anything you want. The (much) harder part is deciding what's relevant to monitor, and what your alarm thresholds should be set at. At $PREVIOUSEMPLOYER, we used Nagios to monitor ~4,000 hosts and about 8,000 to 10,000 services before we started running into scaling problems on a single box. Not quite at 4000 hosts, but we are using Nagios with the nsca client. Works great. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Recommended VOIP Monitoring Tools
Thanks whoever is running an auto response ticket system! Look forward to getting more spam from you! Sent from BETA iOS6 On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On 12-07-13 08:37 AM, Mike wrote: On 12-07-13 06:00 AM, Elliot Murdock wrote: Hello, Which tools are recommendable for monitoring VOIP, bandwidth, server alarms, etc.? Nagios (http://www.nagios.org/) can be configured to monitor pretty much anything you want. The (much) harder part is deciding what's relevant to monitor, and what your alarm thresholds should be set at. At $PREVIOUSEMPLOYER, we used Nagios to monitor ~4,000 hosts and about 8,000 to 10,000 services before we started running into scaling problems on a single box. Not quite at 4000 hosts, but we are using Nagios with the nsca client. Works great. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] New router, registration problems
Linksys firmware? I've had issues with older firmwares and VoIP Sent from my iPhone 4S On Feb 11, 2012, at 1:05 PM, David Woodfall d...@dawoodfall.net wrote: On (17:38 11/02/12), David Woodfall d...@dawoodfall.net put forth the proposition: On (16:48 11/02/12), David Woodfall d...@dawoodfall.net put forth the proposition: I just set up a WRT54GS and now I can't dial out or in. sip show registry shows: CODE: SELECT ALL Hostdnsmgr Username Refresh State Reg.Time draytel.org:5060N x 120 Request Sent I seemed to recall that running in cli always showed a response back, but there's nothing now. Using 1.8.9.2. I have my number at draytel set up to dial my mobile if asterisk is down and it keeps doing it as if it's down. I setup my server as DMZ in the router, as my old one was. Tried with firewall off. Any ideas? I just had a look at debug info and when I dial out I get a busy/congested status back. I can see registration packets going out but no replies. Well I'm not sure why but I just stopped asterisk for a few minutes and then restarted it and now it registers -- _ -- 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] Virtual Server
I run two off virtuozo vps boxes - but capacity will always be the defining value Sent from my iPhone 4S On Feb 10, 2012, at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- 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] India Telecom regulations
Right check out Cordia.LT Sent from my iPhone 4S On Dec 19, 2011, at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: On Tuesday 20 Dec 2011, Steve Edwards wrote: On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? The only thing that is not permitted is bridging Internet calls with the Indian PSTN. In fact, that too is allowed if you have a VoIP licence from the government. Apart from that, as long as you continue using it within your own organisation, any protocol is fine. IANAL. TINLA. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] MySql Custom CDR issues
Are you using FreePBX or another packaged Asterisk? Sent from my iPhone 4S On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote: hello , I have been working hard to solve the issue of custom CDR in the Asterik with Mysql but in vain. I searched google for complete 2 hours but in vain. What i want to achieve is CDR(customcolumn)=anyvaluealthough we can achieve it through other ways like making a script that runs when a call ends and modify the cdr and insert in custom value BUT is there any way to make this work ? Thank you 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] Becoming a CLEC
Agreed. And facilities based CLEC even scarier. Regulatory / billing / PUC legals etc ugh Sent from my iPhone 4S On Nov 14, 2011, at 8:33 PM, Alex Balashov abalas...@evaristesys.com wrote: Worst reason to become a CLEC: improved cost structure. Or, to be precise, it is a counterfactual reason, because it does not result in improved cost structure. This idea is driven by an incomplete understanding of what being a CLEC entails, or, for the less critically thoughtful, the free lunch fallacy. There is no free lunch. There is no such thing as an easy-peasy regulatory reclassification that gets you the same stuff you were paying before, but more cheaply. Becoming a CLEC is a totally different business model than the one you're in, and it entails magnitudinally more technological and regulatory complexity. It's really almost a different vertical. You should become a CLEC only if you want to become a CLEC, not if you want to be an ITSP with a lower cost basis, because you won't be. It is a very capital-intensive, non-trivial endeavour with high barriers to entry for a good reason. There will be people out there who will tell you that those barriers are low; they are on the bridge of failing CLECs, treading water. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
Re: [asterisk-users] Becoming a CLEC
Wow so I left before the end of resale Verizon UNE then. We ran Lucent 5E and Nortel DMS and provided facilities voice and DSL. Having a large SONET fibre infrastructure helped too. Sent from my iPhone 4S On Nov 14, 2011, at 8:53 PM, Alex Balashov abalas...@evaristesys.com wrote: On 11/14/2011 08:36 PM, Robert-IPhone wrote: Agreed. And facilities based CLEC even scarier. I'm curious what sort of thing would be considered a non-facilities based CLEC, since UNE-P was cancelled in 2003. There are some non-interconnected CLECs out there that exist for the sole purpose of leveraging rights of way and stuff like that, but there's not too many things you can do switchless, muxless, DACS-less and not interconnected these days. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
I am adding dickish to my dictionary - thats a hot one! Sent from my iPhone On Sep 25, 2011, at 4:41 PM, Alex Balashov abalas...@evaristesys.com wrote: On 09/25/2011 02:23 PM, Bruce B wrote: Stop wishing for that. I like Asterisk and I will raise a voice when I feel uncomfortable with changes. You won't get an audience if the way you go about it is dickish. You're being a dick, and you know you're being a dick. You're just unwilling to admit it or intellectually engage with that. If you were earnest and sincere about your desire to contribute constructive criticism and effectuate change, you wouldn't start the thread with a sarcastic subject line like Who is the 'creative' mind behind changing Asterisk commands at CLI? That has a mocking, derisive inflection, and you know it has a mocking, derisive inflection. If you expect to be taken seriously, you need to align your behaviour with your stated objective--unless that's not actually your objective, and in fact your objective is to be an inflammatory jerk. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
Re: [asterisk-users] Question about voip.ms service.
I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone On Sep 13, 2011, at 5:14 PM, Danny Nicholas da...@debsinc.com wrote: That’s what this part of extensions.conf should do: ; inbound context example for your DID numbers, do not add the number 1 in front [voipms-inbound] exten = 7863643011,1,Answer() ;your DID From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Yup, that part I got. What I am not clear about is how to set up the DID to go to my URI. When I select manage DIDs and click on the one I want to change, I see the following options for routing the DID x SIP/IAX - [main account] IAX2/10 - with my account number x SIP URI - SIP:mysi...@myuri.com:5060 x System - Hangup There are several other options but they are not selectable for me because I have not set up to use them. I used to have the routing set to SIP URI where I was able to specify my URI where the call was routed to. But with the SIP/IAX option I do not have that ability. I am missing something fundamental here. My asterisk has the iax.conf and extensions.conf entries ready to receive calls from voip.ms, but I don't know how to tel voip.ms to send the calls to my asterisk with the IAX protocol. I understand this is probably a question for the voip.ms folks, but since a couple of people mentioned earlier that they were rocking with IAX, I thought it would be an easy question for them to point me in the right direction. Thanks. On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel dai...@pervasivetelecom.com wrote: I was lurking in this conversation and I went to look more carefully at the voip.ms site. I found sample files at http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 Hope that helps. On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote: I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that
Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
Asterisk is a company? This is news to me Sent from my iPhone On Sep 12, 2011, at 5:35 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: See comments inline. On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. If you want an OS that is going to be supported a year from now, don't use Fedora. Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much beta RHEL. It's EOL is one year from my understanding. You want to install the very minimum as most people would agree, why do you think you need a GUI. Best practice is to only install the bare minimum on a server. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? It has and will cause issues. I have installed KDE or whatever but booted to init 3 for a couple of machines. I could go to init 5 if I had to, but I never did had to. I don't see a single pro, but there are many cons. What benefit do you get from KDE? Why do you want it. Is this just going to be an asterisk server or a desktop? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. How does remote desktop help you over an SSH CLI? FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. Ok, I can understand, I used to be like this for a while. I am a huge fan of Webmin for a GUI. It allows for almost everything and for me, it is better than KDE or anything else. It is just a webpage with tools attached. No big potential problem there. I look forward to your input. Thanks I have been using Vyatta (paid for with phone support.) It makes for the most powerful Asterisk platform you can imagine. There is a learning curve but I love what I have put together. There are howtos everywhere and if you buy licenses, you get excellent support and online training courses. It is a very firewall/Router. It handles everything from OpenVPN, awesome security features, IPS, and even QoS, wireshark. I put webmin and NTOP on these machines as well. Vyatta has become my new platform for Asterisk. Check it out http://www.vyatta.org/documentation There is very little you cannot do, but don't have to use the features if you don't want to. Vyatta is also a company like Asterisk. Vyatta is the baby of former bigtime corporate Cisco guys. Asterisk is the baby of former Adtran execs. Thanks, Steve T Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Phone numbers and asterisk
what do you mean? Like speed dial or directory? Sent from my iPhone On Sep 4, 2011, at 6:47 PM, neo haux neo.h...@gmx.com wrote: Hi, It is possible to save all the phones numbers on asterisk servers instead of doing so manually in each VoIP device ? Does SIP take care of such configuration ? Thanks -- _ -- 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] sip attacks
hard to equate sip attack to ping performance.. Run mtr for a bit. Also try tcpdump or wireshark or tethereal. If you are really paranoid recycle all your passwords Sent from my iPhone On Jul 31, 2011, at 7:04 PM, Dave George dgeo...@teletoneinc.com wrote: My asterisk server is getting bogged down every 5 minutes. My ping time is going from 60ms to 800 ms and the call quality is bad. I have fail2ban running and I am using iptables. I have two ip connections to the box. How can I tell if the poor performance is due to sip attacks? I don't see any reg attempts in my asterisk cli. I use to get frequent attacks but fail2ban seems to be taking care of that. See how ping time gets worst in a short space of time and server performance at the time: 64 bytes from 4.2.2.1: icmp_seq=6 ttl=55 time=87.8 ms 64 bytes from 4.2.2.1: icmp_seq=7 ttl=55 time=99.8 ms 64 bytes from 4.2.2.1: icmp_seq=8 ttl=55 time=107 ms 64 bytes from 4.2.2.1: icmp_seq=9 ttl=55 time=115 ms 64 bytes from 4.2.2.1: icmp_seq=10 ttl=55 time=120 ms 64 bytes from 4.2.2.1: icmp_seq=11 ttl=55 time=122 ms 64 bytes from 4.2.2.1: icmp_seq=12 ttl=55 time=123 ms 64 bytes from 4.2.2.1: icmp_seq=13 ttl=55 time=126 ms 64 bytes from 4.2.2.1: icmp_seq=14 ttl=55 time=122 ms 64 bytes from 4.2.2.1: icmp_seq=15 ttl=55 time=142 ms 64 bytes from 4.2.2.1: icmp_seq=16 ttl=55 time=142 ms 64 bytes from 4.2.2.1: icmp_seq=17 ttl=55 time=137 ms 64 bytes from 4.2.2.1: icmp_seq=18 ttl=55 time=186 ms 64 bytes from 4.2.2.1: icmp_seq=19 ttl=55 time=255 ms 64 bytes from 4.2.2.1: icmp_seq=20 ttl=55 time=310 ms 64 bytes from 4.2.2.1: icmp_seq=21 ttl=55 time=387 ms 64 bytes from 4.2.2.1: icmp_seq=22 ttl=55 time=445 ms 64 bytes from 4.2.2.1: icmp_seq=23 ttl=55 time=514 ms 64 bytes from 4.2.2.1: icmp_seq=24 ttl=55 time=583 ms 64 bytes from 4.2.2.1: icmp_seq=25 ttl=55 time=650 ms 64 bytes from 4.2.2.1: icmp_seq=26 ttl=55 time=715 ms 64 bytes from 4.2.2.1: icmp_seq=27 ttl=55 time=783 ms 64 bytes from 4.2.2.1: icmp_seq=28 ttl=55 time=821 ms 64 bytes from 4.2.2.1: icmp_seq=29 ttl=55 time=810 ms 64 bytes from 4.2.2.1: icmp_seq=30 ttl=55 time=832 ms 64 bytes from 4.2.2.1: icmp_seq=31 ttl=55 time=812 ms 64 bytes from 4.2.2.1: icmp_seq=32 ttl=55 time=821 ms 64 bytes from 4.2.2.1: icmp_seq=33 ttl=55 time=826 ms 64 bytes from 4.2.2.1: icmp_seq=34 ttl=55 time=815 ms 64 bytes from 4.2.2.1: icmp_seq=35 ttl=55 time=821 ms 64 bytes from 4.2.2.1: icmp_seq=36 ttl=55 time=824 ms top - 19:02:38 up 4 days, 11:26, 4 users, load average: 0.36, 0.75, 0.82 Mem: 4051312k total, 1062964k used, 2988348k free, 167004k buffers Swap: 6094840k total,0k used, 6094840k free, 680144k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 4245 root 15 0 791m 86m 10m S 39.6 2.2 1192:32 asterisk 18280 root 15 0 3812 600 516 S 2.0 0.0 0:59.00 pppoe 2582 root 15 0 5912 628 504 S 0.3 0.0 2:02.19 syslogd 18978 root 15 0 12744 1096 812 R 0.3 0.0 0:00.02 top 1 root 15 0 10352 700 588 S 0.0 0.0 0:01.14 init 2 root RT -5 000 S 0.0 0.0 0:00.01 migration/0 3 root 34 19 000 S 0.0 0.0 0:31.90 ksoftirqd/0 4 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/0 5 root RT -5 000 S 0.0 0.0 0:00.01 migration/1 6 root 34 19 000 S 0.0 0.0 0:08.43 ksoftirqd/1 7 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/1 8 root RT -5 000 S 0.0 0.0 0:00.13 migration/2 9 root 34 19 000 S 0.0 0.0 2:40.56 ksoftirqd/2 10 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/2 11 root RT -5 000 S 0.0 0.0 0:00.05 migration/3 12 root 34 19 000 S 0.0 0.0 0:44.56 ksoftirqd/3 13 root RT -5 000 S 0.0 0.0 0:00.00 watchdog/3 14 root 10 -5 000 S 0.0 0.0 0:00.02 events/0 15 root 10 -5 000 S 0.0 0.0 0:00.00 events/1 16 root 10 -5 000 S 0.0 0.0 0:00.00 events/2 17 root 10 -5 000 S 0.0 0.0 0:00.00 events/3 18 root 10 -5 000 S 0.0 0.0 0:00.00 khelper 55 root 10 -5 000 S 0.0 0.0 0:00.00 kthread 62 root 10 -5 000 S 0.0 0.0 0:00.07 kblockd/0 63 root 10 -5 000 S 0.0 0.0 0:00.01 kblockd/1 64 root 10 -5 000 S 0.0 0.0 0:00.00 kblockd/2 65 root 10 -5 000 S 0.0 0.0 0:00.00 kblockd/3 66 root 17 -5 000 S 0.0 0.0 0:00.00 kacpid 166 root 17 -5 000 S 0.0 0.0 0:00.00 cqueue/0 167 root 18 -5 000 S 0.0 0.0 0:00.00 cqueue/1 Dave -- _ --
Re: [asterisk-users] Lightning and thunder
gerbals Sent from my iPhone On Jul 27, 2011, at 5:32 PM, Hans Witvliet h...@a-domani.nl wrote: On Wed, 2011-07-27 at 09:44 -0400, Claude Hayn wrote: We are frequently losing power during lightning storms. (Yes we have UPS, but often by the time power comes back up the UPS has run out of juice) snip Does anyone know of a solution for this issue? Having to get up in the late night to manually reboot the Asterisk box is getting old! Perhaps an other suggestion... Re-install asterisk on a other piece of hardware. There are small boxes that consume less than 5 Watt. If you put that on your UPS, it will last longer. Other one, ever thought of an alternative power source? Either solar of conventional? 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 -- _ -- 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] Securing Asterisk
Such a pointless argument. The same problem can happen on any voip platform including freeswitch. Again it's a knowledge thing. BTW if you were paying attention to your logs or practiced good admin skills you would have seen the attacks and stopped them. I swear by fail2ban and other hardening techniques. If you honestly think you can just run the box out in the open after running a yum / apt or rpm command you are in the wrong position. Know this is going to sound harsh but you deserve the pay cut if not termination. Sent from my iPhone On Jul 23, 2011, at 2:13 PM, Danny Nicholas da...@debsinc.com wrote: Simple economics tells me that we can't pay enough guys $X U.S. to stop the problem when we are competing with multiple folks working for $0.X US. Asterisk isn't the problem, it's just another limb on the victim tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Saturday, July 23, 2011 1:10 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Securing Asterisk On 11-07-23 01:38 PM, CDR wrote: I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user information, and also, b) disable any response to any REGISTER packet altogether. Can somebody please write patch? Or should we go broke trying to stop the flood of criminals coming from abroad? Federico I'm not sure I understand your statement. Because your customer was hacked for $50,000 and your pay was cut in half, it is a result of Digium (or the Asterisk project) 'hiding from the real world'? Your previous point aside, may I ask how your client solved the problem? I'm assuming they are still operating an Asterisk box without the patches you have requested. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 -- _ -- 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] Requires
wrong address - but I can come Monday if you like ;) Sent from my iPhone On Jul 16, 2011, at 8:58 AM, mahesh katta maheshka...@flexydial.com wrote: Dear Ashirwad, Please make ready below things for demo in pune .MONDAY needs to be ready for test in our office. 1. PRI card single span 2. PRI cable 3. Server 4. SIM cards 4 with recharge. Best Regards, Mahesh Katta BUZZWORKS Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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
Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support
+1 for Xen -1 for VB Sent from my iPhone On Jul 8, 2011, at 10:00 PM, Doug Lytle supp...@drdos.info wrote: Warren Selby wrote: Not trying to start a war here, That may be, but I have experience with VB. 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] Cisco IP Phones and Skinny in asterisk
wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Cisco IP Phones and Skinny in asterisk
You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) Sent from my iPhone On Jun 20, 2011, at 6:44 PM, bilal ghayyad bilmar...@yahoo.com wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Where do u download the SIP firmware usually for your Cisco IP Phones? Your kindly help is highly appreciated. Regards Bilal --- I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, -- _ -- 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] Ground Start ATA / VOIP Gateway
considering providing the sip trunking nyself via asterisk. the sip trunking looks expensive - card and licenses from nec. Sent from my iPhone On Jun 14, 2011, at 6:06 PM, John Novack jnov...@stromberg-carlson.org wrote: that system can also handle IP trunks, though the equipment might not be available to you or outside your budget window How does this relate to Asterisk, or does it? John Novack Robert Huddleston wrote: I’ll have to look at that then – as I thought the card actually said “Ground Start” on it.. I may have missed or it was scratched off the word loop start From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 5:20 PM To: Robert Huddleston Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway The SV8100 can do either ground or loop Assuming you can access the system it can easily be changed. Programming manual here: http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming Manual_1.pdf the original installer may have locked it down, but it CAN be changed. John Novack Robert Huddleston wrote: Ya – customer is on a nice NEC SV8100.. The card is a ground start card.. they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown cross-connect. But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and want to use Ethernet for wan… So IAD2431 would be great – but if it only allows T1/E1 for WAN – I’m shot. From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Robert Huddleston Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway Robert Huddleston wrote: Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 – but okay if not.. I don't know of any ATA that will do GS An RJ-21 is the designation for a 66 block with 25 pair connector on the side GS is available with many channel banks though a T1 card and channel bank might be overkill for your application. Is this to go into a legacy switch? Most have line cards that can be easily switched to Loop Is this in the US, or ??? John Novack -- Dog is my Co-pilot -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ground Start ATA / VOIP Gateway
exactly my other concern - can just drop sip card in and put on the net - would also have to get an sbc - which would be more than an ATA. considering just using a cisco router (low end XM) and throwing a high density voice card in it Sent from my iPhone On Jun 14, 2011, at 6:48 PM, John Novack jnov...@stromberg-carlson.org wrote: Agreed NEC isn't cheap. Their products are generally pretty good and robust though. I have an earlier one still working for 18 years and counting Of course, when one considers the asterisk machine, configuration time, firewall and the rise in sip hacking sip trunking can easily turn into a PITA. John Novack Robert-iPhone wrote: considering providing the sip trunking nyself via asterisk. the sip trunking looks expensive - card and licenses from nec. Sent from my iPhone On Jun 14, 2011, at 6:06 PM, John Novack jnov...@stromberg-carlson.org wrote: that system can also handle IP trunks, though the equipment might not be available to you or outside your budget window How does this relate to Asterisk, or does it? John Novack Robert Huddleston wrote: I’ll have to look at that then – as I thought the card actually said “Ground Start” on it.. I may have missed or it was scratched off the word loop start From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 5:20 PM To: Robert Huddleston Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway The SV8100 can do either ground or loop Assuming you can access the system it can easily be changed. Programming manual here: http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming Manual_1.pdf the original installer may have locked it down, but it CAN be changed. John Novack Robert Huddleston wrote: Ya – customer is on a nice NEC SV8100.. The card is a ground start card.. they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown cross-connect. But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and want to use Ethernet for wan… So IAD2431 would be great – but if it only allows T1/E1 for WAN – I’m shot. From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Tuesday, June 14, 2011 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Robert Huddleston Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway Robert Huddleston wrote: Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 – but okay if not.. I don't know of any ATA that will do GS An RJ-21 is the designation for a 66 block with 25 pair connector on the side GS is available with many channel banks though a T1 card and channel bank might be overkill for your application. Is this to go into a legacy switch? Most have line cards that can be easily switched to Loop Is this in the US, or ??? John Novack -- Dog is my Co-pilot -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
I also had trouble w/ these phones at first. There was a DHCP option (?81?) you'll have to google it. The phones would not talk to tftp until I set dhcp option. The console aux cable is easy to build and VERY useful Sent from my iPhone On Jun 13, 2011, at 8:31 PM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Bilal, I suggest you turn on logging on your tftp server to see what files are actually being requested, and if the the tftp server is dishing them out... Try adding a few v's to your tftp setup: File: /etc/xinetd.d/tftp Line to change: server_args = -s /tftpboot -v -v -v Look in /var/log/messages for the output. Also, I believe your 7942G has a console/aux port which is a serial port, you can learn what is happening as the phone boots up with that too. Good Luck! Mark On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote: Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- 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
