November 2009 - VoiceOps - lists.voiceops.org

Cisco 7940 PTT
by ujjval＠simplesignal.com Nov. 16, 2009

Nov. 16, 2009

Can a Cisco 7940/7960 do Push to Talk..any config suggestions?

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Global NAPS didn't file number audits
by peter＠4isps.com Nov. 13, 2009

Nov. 13, 2009

GLOBAL NAPS CALIFORNIA, INC. OCN# 5300. Notified Global NAPs California, Inc., of its apparent liability for a forfeiture in the amount of $80,000 for willfully and repeatedly violating the sections 52.15(k)(1) and 52.15(f) of the Commission's rules, and Commission orders. Action by: Chief, Enforcement Bureau. Adopted: 11/10/2009 by NALF. (DA No. 09-2375). EB http://hraunfoss.fcc.gov/edocs_public/attachmatch/DA-09-2375A1.doc http://hraunfoss.fcc.gov/edocs_public/attachmatch/DA-09-2375A1.pdf http://hraunfoss.fcc.gov/edocs_public/attachmatch/DA-09-2375A1.txt

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PSTN connection to SONUS
by jvanick＠spruce.oaknet.com Nov. 12, 2009

Nov. 12, 2009

Actually you'll probably want to use a PRI config ISUP is used if you're connecting to a switch that requires SS7 signalling. as far as I'm aware, asterisk doesn't support SS7. you'll need to use the Sonus EMS Task Configurator to build a PRI trunk group this should be all covered in the online help. Make sure you set the "Switch Side Behavior" to Network and set a switch type that matches what you set the Asterisk configuration to. --J > > Hi Jason=20 > > Thanks for the information that you gave. I want to connect to an > Asterisk PBX which will act as the termination. So I guess I should be > using an ISUP connection. Also I will be using an E1 to connect the > same. > > Warm Regards > Venugopal G > ************************************************************************ > ************************************************************************ > ************************************************* > =20 > > -----Original Message----- > From: Jason Vanick [mailto:jvanick at spruce.oaknet.com]=20 > Sent: Thursday, November 12, 2009 8:01 PM > To: Gopalakrishnaiyer Venugopal-Q16770 > Subject: Re: [VoiceOps] PSTN connection to SONUS > > There are different procedures based on what kind of trunk you want to > add... > > is it ISDN PRI, Is it ISUP, is it CAS etc...=20 > > What kind of circuit are you planning? DS1, DS1 on DS3, or optical? > > this should be all covered in the manuals. > > It's not an extremely involved, but you do need go get the steps done in > the correct order. > > -J > > > I have a SONUS GSX9000 chassis.I want to add PSTN connection to this=20 > > SONUS.I am new to the switch background.Does anyone have the procedure > > > to add a PSTN connection to the SONUS Switch > >=20 > >=20 > > Warm Regards > > Venugopal G >

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Billing of Forwarded Calls
by scott＠sberkman.net Nov. 11, 2009

Nov. 11, 2009

So how do most of you deal with billing of forwarded calls (specifically where the calling number on the forwarded leg is using the original calling number from the inbound leg) in a SIP environment when the originally called number is not preserved in the new invite? In this case there is no way to match the calling or called number to a specific customer. Do you bill by IP address or interface instead? Do you somehow use a system that correlates the forwarded leg to the original inbound leg? I've come across this issue a few different times when trying to bill off of SIP messaging logs, for instance radius off a SIP SBC or SQL logs from SER. Thanks, -Scott

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GLBX Issues Chicago
by Shri＠pressone.net Nov. 11, 2009

Nov. 11, 2009

Anyone seeing any termination issues on Global Crossing network? we have DID's in Chicago that have no inbound. Thanks shri

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Load balancing
by anorexicpoodle＠gmail.com Nov. 11, 2009

Nov. 11, 2009

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

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BYOB CPE
by robert.dawson＠mindshift.com Nov. 11, 2009

Nov. 11, 2009

What is the general consensus on CPE for hosted VoIP customers that are providing their own connectivity. Are you using whatever the customer provides, requiring a particular device, etc.? We typically use Edgewater, just wanted to see what others were doing. Robert Dawson - MCSE, MCTS Microsoft Exchange 2007 NOC 3/VoIP Engineer mindSHIFT Technologies, Inc. 3975 Fair Ridge Drive, Suite 200 South - Fairfax, VA 22033 Tel: 571-432-4010 Cell: 571-234-2621 www.mindSHIFT.com <http://www.mindshift.com/> Ranked #1 in managed IT services by MSPmentor(r) Boston - New York City - Philadelphia - Washington DC We make IT work for your business(r) P Please consider the environment before printing this e-mail

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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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