August 2011 - VoiceOps - lists.voiceops.org

Broadworks conferencing
by RDawson＠alliedtelecom.net Aug. 30, 2011

Aug. 30, 2011

For the folks running Broadworks, are you using the native conference application or a third party solution? If third party, hosted or in-house. What drove your decision? Thanks, Rob

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syniverse question
by zavoid＠gmail.com Aug. 25, 2011

Aug. 25, 2011

Hey all, If you are connected with syniverse for SMS, what are the monthly minimums they have for you? I'm getting a quote of $7,500 monthly minimums right now. thanks, Colin

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Small SS7 STPs
by PChilds＠internode.com.au Aug. 25, 2011

Aug. 25, 2011

Gday folks. Looking for some options for small sized SS7 ITPs. Low speed (ie timeslots in a 2mb circuit) and m3ua -- pretty much like the Cisco 2811 ITP (before it was EOL'd). Need to handle a handful of old TDM signalling links into our M3UA softswitches. Currently I'm running 4 geographically dispersed Cisco 2811 ITPs with about 4 low speed links each -- this works well for us, but interested in anything else out there since the 2811 will soon be end-of-sale. Cheers, Peter

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DC Earthquake
by zavoid＠gmail.com Aug. 25, 2011

Aug. 25, 2011

Well I didn't feel the quake here in NJ(although others did). I can't reach anyone on AT&T's wireless network(and i could before i swear!) in NJ or D.C. Anyone else having similar issues? Colin

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Qwest Wholesale SIP (Toll-Free Inbound) Salesperson
by gmina＠connectfirst.com Aug. 25, 2011

Aug. 25, 2011

Anyone have contact information for someone at Qwest in the Toll-Free SIP side of things? Their online sales inquiry page is not doing its job. Thanks, Geoff

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ACME management interface
by ujjval＠simplesignal.com Aug. 23, 2011

Aug. 23, 2011

Hi We are seeing acme management interface timing out for SSH via a cisco pix firewall. Has anyone encountered ssh issues with Acme mgmt interface in general wamcom0? Ujjval K

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VoIP Abuse to Twitter
by sil＠infiltrated.net Aug. 22, 2011

Aug. 22, 2011

Apologies for cross posting but some of us aren't on the other list (vice/versa) and thought both groups would benefit. For those familiar with the VoIP Abuse Project, no need to explain the gist of this. I got tired of parsing through the alerts (lists) I receive via email daily. They're long and sometimes I don't have the time to post them all. So for now, posting VoIP Abuse addresses straight to Twitter. So, anyone trying to compromise a pbx, is now autoposted on an hourly basis to Twitter. Still working on pulling, have about 4 machines linked up now, will mop em up during the week. http://twitter.com/#!/voipabuse Now, you can concoct a quick script off of it, e.g.: links -dump "http://twitter.com/voipabuse"|awk '/attacker/{print "iptables -A INPUT -s "$2" -j DROP"| "sort -u"}' Will get a quickie soon from my Acme's, nCites, etc. when I have time. For those NOT familiar with it, please Google it as I don't feel like typing anymore ;) (sorry) -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x2BF7D83F210A95AF

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T-Mobile goes straight to voicemail
by d＠d-man.org Aug. 22, 2011

Aug. 22, 2011

Hello, We've discovered that our VoIP calls, via provider X, go directly to voicemail for any T-Mobile customers called. Calls via Skype ring the phone, as do landlines. This is reproducible ? happens every time. Anyone seen this & have any ideas on the cause? I've confirmed we're setting Caller ID and ANI information correctly on our side, I believe anyway. But generally speaking how would a call even terminate at T-Mobile but not ring the phone? Seems od? -- [cid:85F2F091-9DE6-4FBD-B9CF-56F9E8CF03C2] Darren Schreiber CEO / Co-Founder visit: www.2600hz.com<http://www.2600hz.com/> sip:darren at 2600hz.com tel: 415-886-7901

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Panasonic 600/620 - SIP trunk?
by graham.freeman＠cernio.com Aug. 19, 2011

Aug. 19, 2011

Howdy, folks, I'd like to set up a SIP trunk with an existing Panasonic TDE 600/620 on one end and my fav conf call service (ZipDX) on the other end. Preferably this would be with g.722, but g.711 would be acceptable. Whenever I mention SIP to the Panasonic VAR, he quickly corrects me to say that it's "Panasonic proprietary VOIP" (as if that's a good thing). I'm not 100% convinced that it's not SIP. I could (and will) throw Wireshark on the network span that currently bridges two of these PBXes, but I'd really like to hear from someone who has done this in real life -- gotchas being what they are. Any tips? On- or off-list is good. Voice is, ironically, not my preferred mode of communication for this, but I'll take free advice however I can get it. thanks! Graham

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RFO: Cisco ASR platform as an SBC
by anorexicpoodle＠gmail.com Aug. 18, 2011

Aug. 18, 2011

All, I was wondering who out there has actually used Ciscos ASR platform as an SBC and what are your opinions of it? I am well acquainted with the offerings from Acme and Sonus but I have a specialized need that those vendors may be pricing themselves out of in licensing, but it MUST be a B2BUA which eliminates OpenSIPS/Kamailio as a potential solution. I'm also looking for a solution that can be scaled out to between 16k/32k sessions/node with high availability and media handling, but with minimal signaling mangling and transformations. Please feel free to share your experiences good, bad or otherwise. Thanks in advance! -Ryan

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