- VoiceOps - lists.voiceops.org

No audio issues
by ujjval＠simplesignal.com Nov. 9, 2009

Nov. 9, 2009

Anyone experiences no audio, calls not connecting over Level 3? Ujjval Karihaloo VP Voice Engineering SimpleSignal Inc. IP Phone: +13032428610 Sent from my iphone

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Hackin attempts to Broadsoft voice portal
by ujjval＠simplesignal.com Nov. 9, 2009

Nov. 9, 2009

Lot of failed attempts from 1 particular number.... 15096346223<tel:+15096346223>. Seems to be scanning thru voice mail boxes serially attempting passwds & locking them out. Last time v saw this i tracked this to be a magic jack number, i called back & a person answered & said that he can fix my voice mail for me if i gave him my password. I emailed magicjack support & they never responded. Is there anything we can do to bring these malicious hackers to justice? Any one has experience un this area legally given that these hackers are mostly outside the US. Ujjval Karihaloo VP Voice Engineering SimpleSignal Inc. IP Phone: +13032428610 Sent from my iphone

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Free VoIP/Asterisk security webinar - FBI, VOIP SA, Digium
by jtodd＠digium.com Nov. 9, 2009

Nov. 9, 2009

In the wake of genuine concerns (and a lot of FUD) that has recently swirled around the question of VoIP security, Digium has decided to put together a free webinar with participants from the FBI, the VoIP Security Alliance, and Digium training staff. It will cover some general basics of the problem set as discussed by the FBI and VOIP-SA representatives, as well as specific methods and techniques you might be able to use to better-protect yourself if you're running Asterisk or Switchvox (Digium's commercial Asterisk PBX) platforms. I'm certain anyone running any type of VoIP platform will be able to benefit from the session, even if the Asterisk-specific discussion and defense mechanisms do not exactly apply to your circumstances. The participants of this list (voiceops) are possibly not in the exact demographic of the user community that we're trying to serve, as you are mostly senior engineering folk versus the mid-level or beginner administrators and managers we're hoping will attend. However, it still may be interesting and is operationally relevant to many of you, and certainly can be passed along to others to whom it may be new information. Feel free to pass this announcement around to other lists or distributions which you think would be appropriate. Webinar: Asterisk VoIP Security Time: Friday, November 13, 2009 10:30 AM - 11:30 AM CST (+30m Q+A) Sign up: http://bit.ly/24L4Qi JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/

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International SIP Terminations
by Shri＠pressone.net Nov. 9, 2009

Nov. 9, 2009

I'm looking to provide some ultra cheap international calling to Africa and Latin America. Does anyone have a contact for a "lower tier" international Sip term provider? Thanks Shri

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What DNS Is Not
by abalashov＠evaristesys.com Nov. 8, 2009

Nov. 8, 2009

I thought this was a rather thought-provoking article, especially in light of the ways in which DNS is frequently abused in VoIP relative to the intentions of its original design: http://queue.acm.org/detail.cfm?id=1647302 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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Load balancing
by bsneddon＠gc1.com Nov. 6, 2009

Nov. 6, 2009

I'll chime in with at least one potential option. If the free SIP proxies/load balancers all fell off the table solely because of lack of stateful HA, then perhaps running your proxy/proxies on VMware vSphere and taking advantage of their Fault-Tolerance feature is an option. I haven't used it personally quite yet, however it's supposed to provide stateful failover for the entire system. Brian ----- "anorexicpoodle" <anorexicpoodle at gmail.com> wrote: > I have come across the need for a stateful SIP load balancer with > fail-over and HA capabilities, and I am hoping some on this list can > chime in with some personal experience. > > The need is not for a particularly large volume of traffic, more to > act > as a socket to adapt my active/active model UM system to Metaswitch's > 1:1 UM system model, which cannot support active/active application > servers. Yes I felt this was a strange omission in functionality as > well. > > I have pretty much ruled out OpenSER/OpenSIPS/Kamilio because it > isn't > stateful in HA failover. Passing it through an Acme is ridiculously > costly for the need i have, and most hardware based load balancers I > am > finding just aren't sip-aware, so I don't see them doing much better > than OpenSER in a fail-over scenario, it would just be a different > kind > of ugliness. > > Thanks in advance > > > _______________________________________________ > VoiceOps mailing list > VoiceOps at voiceops.org > https://puck.nether.net/mailman/listinfo/voiceops

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Anyone getting hammered by 512-687-1442?
by csteinman＠rgts.com Nov. 6, 2009

Nov. 6, 2009

Has anyone gotten hammered from calls originating from 512-687-1442? We received 391 calls in approx 90 seconds from that single number doing a sweep of numbers to one of our exchanges. Thanks, Charles Steinman Sr. Network Engineer RGTS 1221 Avenue of the Americas New York, NY 10020 Phone: 212-282-2249 http://www.rgts.com<http://www.rgts.com/>

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DST handling
by hiersd＠gmail.com Nov. 6, 2009

Nov. 6, 2009

Do you do anything special for the DST changes? In other words, do you try to adjust for the fact that the 0200 hour occurs twice in the fall and that the 0200 hour is completely missing in the spring on reports and such? Thanks, David

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full to compact header conversion
by hiersd＠gmail.com Nov. 5, 2009

Nov. 5, 2009

Anyone seen a slick way to convert headers from full to compact? After a bunch of diversions, we get close to the PMTU. Thanks, David

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DASH issues
by justin＠justinshore.com Nov. 5, 2009

Nov. 5, 2009

Is anyone else running into issues with DASH's SBC changes from this past morning's maintenance window? We're experiencing two new issues. The first is a DTMF issue. We're seeing DASH send us RFC2833 DTMF over the SIP trunk but the MetaSwitch is not liking what the new DASH SBCs are sending. In turn the MetaSwitch is not sending DTMF on to the downstream customer. It sounds like RFC2833 negotiation between the MetaSwitch and their new SBC leaves only ulaw as an option which doesn't appear to agree with MetaSwitch. Thus no DTMF on towards the customer. We were moved back to the old SBC and the DTMF problem has been resolved for now. We've been told that this was an issue earlier with DASH and MetaSwitch and that it was resolved at a particular MetaSwitch customer's site. We're working back channels to try and figure out what that fix was. The second issue is with what initially appears as a no-way audio on calls to certain numbers at different customer sites. When you capture the traffic you realize that there is 2-way audio but it only contains comfort noise. DIDs don't seem to be affected. The only numbers chosen for this particular problem are the main numbers at the sites. The sites in question all use the same remote media gateway model on the MetaSwitch, have Edgemarc CPEs and Aastra phones. The lines are also configured for SIP forking with several phones on the same line. However it doesn't affect all of our customers configured in what appear to be the exact same way. Some work perfectly fine. What we're hearing from DASH is that their upstream is putting the call on-hold for some reason as soon as the receiving end picks up. Very strange. We're troubleshooting that one in the morning. Has anyone else run into issues with the migration to the new SBCs at DASH? Since we only had outgoing calls going over the new DASH SBCs and incoming on the old SBCs these problems weren't ever discovered in testing. Without that there really wasn't any testing done on our end. Thanks Justin

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