November 2009 - VoiceOps - lists.voiceops.org

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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AudioCodes Mediant 2k
by nvermeer＠rgts.com Nov. 4, 2009

Nov. 4, 2009

Has anyone here been able to configure Mediant 2k to send "Calling Party Name" in the ISUP message to the connected ISDN device (in this case an Avaya PBX). This is an IP->Tel call, not a Tel->IP. The SIP UA is a Broadworks platform. Nicholas Vermeer Manager, Engineering (ph) +1.212.282.2512 (fax) +1.212.282.2668 (email) nvermeer at rgts.com<mailto:nvermeer at rgts.com> Rockefeller Group Technology Solutions, Inc. 1221 Avenue of the Americas New York, NY 10020-1095

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PW call failures, outage?
by owen＠impulse.net Nov. 4, 2009

Nov. 4, 2009

I have multiple reports from customers with problems using PacWest -- they're getting an intercept recording of "all circuits busy" albeit on only a handful of destinations, which then quickly transitions to "fixed," by which they mean if you wait through 4-9 seconds of post-dial delay the call completes. Start time of 10:00 am PST, still going on. Anyone else seeing this? -- Owen Roth Network Engineer Impulse Internet Services http://impulse.net/customer-support mailto:owen at impulse.net Tech Support: 805-456-5800 w: 805-884-6332

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t-mobile outage
by jarrod＠fed-com.com Nov. 3, 2009

Nov. 3, 2009

Ive been getting reports from a few customers who cant reach certain cell phones.. They all seem to be t-mobile... I guess they are having a huge outage.. http://news.cnet.com/8301-13860_3-10390097-56.html -- Jarrod Lash, <jarrod at fed-com.com> Federated Communications Office: +1-412-357-2127 Mobile: +1-412-999-0049 Fax: +1-412-545-8368

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