Re: [asterisk-users] Brute force attacks
On Thu, Jul 1, 2010 at 9:16 AM, Ishfaq Malik wrote: > Hi > > We've just noticed attempts (close to 20 attempts, sequential peer > numbers) at guessing peers on 2 of out servers and thought I'd share the > originating IPs with the list in case anyone wants to firewall them as we > have done > > 109.170.106.59 > 112.142.55.18 > 124.157.161.67 > > 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 > We have noticed the same sort of activity on our server. The originating IP addresses attempting access were: 204.9.204.145 (hosted at U.S. Colo, I believe) 91.203.132.149 (Nephax) 130.70.157.186 (University of Louisiana) 61.160.121.46 (Chinanet) 109.170.0.10 (ReasonUP Ltd) -- John Timms IT Department - Gnoso Inc. j...@gnoso.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] Question about MaxRetries in the Asterisk Outgoing folder
I'm doing some automated calling by putting .call files in the Outgoing folder of Asterisk. I'm concerned this might be a stupid question, but I'm pretty sure I've done my research well and I'm unable to come up with an answer on my own. I want to know: what happens to the .call files after the "MaxRetries" number has been reached? In my experience, they stay in the Outgoing folder, but are never deleted. Instead, Asterisk keeps processing them, but never actually making a call. In my mind, once the MaxRetries number has been met, Asterisk should do something to get rid of the files, whether moving them to another "failed" folder or just deleting them. You can see an example of my problem below. The Yellow Highlighted remarks are my own for clarification and are not in the actual .call file. -- Channel: SIP/8644161...@vitel-outbound MaxRetries: 9 <= Set to retry 9 times RetryTime: 120 <= Retrys after 120 seconds Context: autodial Extension: s Priority: 1 CallerID: 8645553190 Set: USERNUMBER=8644161809-JohnTimms Set: DIGITS=8644161809 <= I've tried the "Archive: Yes" option here, but had no change in behavior StartRetry: 2397 1 (1270149233)<= This line & the following are all added by Asterisk EndRetry: 2397 1 (1270149158) StartRetry: 2397 2 (1270149399) EndRetry: 2397 2 (1270149324) StartRetry: 2397 3 (1270149565) EndRetry: 2397 3 (1270149490) StartRetry: 2397 4 (1270149731) EndRetry: 2397 4 (1270149656) StartRetry: 2397 5 (1270149897) EndRetry: 2397 5 (1270149822) StartRetry: 2397 6 (1270150063) EndRetry: 2397 6 (1270149988) StartRetry: 2397 7 (1270150229) EndRetry: 2397 7 (1270150154) StartRetry: 2397 8 (1270150395) EndRetry: 2397 8 (1270150320) StartRetry: 2397 9 (1270150561) DelayedRetry: 2397 8 (1270151821) DelayedRetry: 2397 8 (1270151942) DelayedRetry: 2397 8 (1270152063) DelayedRetry: 2397 8 (1270152184) DelayedRetry: 2397 8 (1270152305) DelayedRetry: 2397 8 (1270152426) DelayedRetry: 2397 8 (1270152547) DelayedRetry: 2397 8 (1270152668) DelayedRetry: 2397 8 (1270152789) DelayedRetry: 2397 8 (1270152910) DelayedRetry: 2397 8 (1270153031) DelayedRetry: 2397 8 (1270153152) DelayedRetry: 2397 8 (1270153273) DelayedRetry: 2397 8 (1270153394) DelayedRetry: 2397 8 (1270153515) DelayedRetry: 2397 8 (1270153636) ------ If anyone can help me out, that would be much appreciated. -- John Timms (864) 416-1809 johngti...@gmail.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] how asterisk knows which context forward the call to?
Your question is a little vague. I assume that you would be looking for the "GoTo" application. The syntax is explained here: http://www.voip-info.org/wiki/view/Asterisk+cmd+goto <http://www.voip-info.org/wiki/view/Asterisk+cmd+goto>Also, you can look on page 426 of the Asterisk book, which is really helpful if you're new to Asterisk. Download it for free from the publisher here: http://downloads.oreilly.com/books/9780596510480.pdf <http://downloads.oreilly.com/books/9780596510480.pdf>John Timms -- John Timms (864) 416-1809 johngtimms (at) gmail (dot) com -- IT Department - Gnoso Inc. john (at) gnoso (dot) com -- Grapedial- Affordable group phone messaging www.grapedial.com john (at) grapedial (dot) com -- On Wed, Feb 17, 2010 at 8:40 PM, Joseph wrote: > Is there any asterisk guru who can explain me how how asterisk knows which > context forward the call to? > > -- > Joseph > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
Thanks for suggestions, everyone- I should have thought about jitter and latency as I began to use up more & more bandwidth. I was concerned that it was a problem with my configuration of Asterisk, but it looks like is really is a bandwidth issue. By the way, Joe- I've been in another situation with my cableco & Asterisk/VoIP (on a business connection!) and would frequently have trouble getting *one* call that sounded good, even though we had several megabits up & down, with no other traffic on the network. Charter's service is horrible- there were several times pinging Google took over 1 second. John Timms On Sat, Nov 7, 2009 at 2:45 PM, John Timms wrote: > Hi. I'm having trouble figuring out why I'm not able to make many > concurrent VoIP calls on my system. I'm not aiming for a huge number, > because I have purposely bought a low powered system, but I would > think that I could get more. Here are the details: > > I have a small-form-factor Asterisk server with an Intel Atom 230 CPU > (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu > Server 9.04 with the default Debian package manager installation of > Asterisk. (version 1.4) > > Here is what is going on: I'm making outgoing calls (with .call files) > via SIP (using Vitelity's service, if anyone wants to know) with about > 55.0 ms latency between my Bellsouth DSL connection & their servers. > I'm using GSM-format prompts with GSM encoding (disallow=all, > allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. > I have a very fast internet connection, so there is still plenty of > bandwidth, and the "top" command shows that Asterisk is only at about > 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will > "skip" occcasionally, but cell phones have perfect quality. > > I don't think that 7 calls is very many, I'll be happy if I can get 10 > good-sounding calls. Can anyone give suggestions? (If this has been > hashed out elsewhere, I'm happy with a link to more information!) > > Thanks. > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
Hi Fred. The NIC chip is a Realtek RTL8101E, on the motherboard. Network is Bellsouth => modem/router => Asterisk Yes, I am using NAT (assuming you mean that the Asterisk server does not have its own public IP address) Endpoints are outside the network, just standard POTS phones. Vitelity is my SIP provider. By "fast" I mean the best Business DSL Bellsouth has to offer: "Up to 6.0 Mbps downstream - Up to 512 Kbps upstream" I've used iftop on my server while running calls, and I'm under 200 Kbps while my calls are running. John Timms On Sat, Nov 7, 2009 at 4:25 PM, Fred Posner wrote: > On Sat, Nov 7, 2009 at 2:45 PM, John Timms wrote: >> I have a small-form-factor Asterisk server with an Intel Atom 230 CPU >> (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu >> Server 9.04 with the default Debian package manager installation of >> Asterisk. (version 1.4) > > What kind of NIC are you using and what's the network config? ie > Bellsouth -> router -> switch -> you > > Are you NAT'd? > > Where are your endpoints connected? (locally, outside?) > >> I have a very fast internet connection, so there is still plenty of >> bandwidth > > what is the specs for "fast"? > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with concurrent VoIP calls
Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection & their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the "top" command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will "skip" occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
