October 2009 - VoiceOps - lists.voiceops.org

Verso Verilink 8100 series
by carlos＠race.com Oct. 23, 2009

Oct. 23, 2009

Has anyone used the verso verilink iad's? what are your thoughts the good the bad the ugly? Carlos Alcantar Race Telecommunications, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.649.3550 x143 F: 650.649.3551

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Toll Fraud
by nvermeer＠rgts.com Oct. 23, 2009

Oct. 23, 2009

I'd like tap the collective knowledge here for some fraud prevention and detection resources. Anyone have any information they can share on common fraudulent calls (i.e. Countries and associated digit strings) What metrics have worked best to match to detect fraud before it becomes a significant cost? What platforms have people used to analyze the raw data? Links or RTFM pointers appreciated :). Nicholas Vermeer (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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Fiber (Dark/Lit) providers in the Dallas/Ft. Worth and Houston areas
by ssrinivasan＠telesphere.com Oct. 23, 2009

Oct. 23, 2009

Folks, I am looking for fiber providers (lit services and dark) in the Dallas/Ft worth and Houston areas. I would appreciate as many names you have worked with or know of. Thanks. SANJAY SRINIVASAN 480.385.7062 (Anywhere) | 480.385.7064 (fax) | ssinivasan at telesphere.com<mailto:ssrinivasan at telesphere.com> | www.telesphere.com<http://www.telesphere.com/> 9237 E. Via de Ventura | Suite 250 | Scottsdale, AZ 85258 [cid:image001.jpg at 01CA53D1.92112D70]

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Level 3 inbound issues
by ujjval＠simplesignal.com Oct. 21, 2009

Oct. 21, 2009

Looks like we are not getting calls delivered to our network by Level 3 area code 973, 561 and some Wyoming TNs... Anyone seeing this? Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg at 01CA5240.0B2921B0]<http://www.simplesignal.com/>

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IPV6
by anorexicpoodle＠gmail.com Oct. 21, 2009

Oct. 21, 2009

Since we, collectively, are steering one of the industries driving up individual utilization of the IPV4 address space as well as being one of the most sensitive to NAT which is the only way through which IPV4 has been sustained as long as it has; it seems like a worthy exercise to discuss our own, and the industries preparedness to adopt IPV6. Has anyone out there had any experience using any of the open source platforms (OpenSIPS, Asterisk, SIPPY etc) with native IPV6? It seems like these projects are the best equipped right now to handle this move since they rely heavily on the network stack of the underlying OS. Are there any endpoints or other CPE that anybody has had luck getting to work over native IPV6? So far I am unaware of any carrier grade (meaning it costs a lot of money) softswtch platforms that are ready for this, or seem like they would be without a multi-year effort. Anybody out there that can enlighten us on this one? Any experience with SBC's or border elements that are or will be ready to handle IPV6 in deployment (either native 6 to 6 or 6 to 4 through a b2bua) Finally any comments or thoughts on the whole topic, I am interested to see other opinions and approaches to the topic.

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ShoreTel PBX interop with Broadsoft Based Switch.
by ujjval＠simplesignal.com Oct. 21, 2009

Oct. 21, 2009

Hi All, We are trying to interop the ShoreTel (behind a Edgemarc - which registers on behalf of the Shoretel as from what I know Shoretel cannot send a SIP register) with our Broadsoft setup..I was wondering if someone has done it. We are trying to get the Shoretel up and running, while I have it configured as per the Broadsoft Partner config guide, the Shoretel system still continues to use Port 5004 for RTP (which is its default behavior when not configured for SIP), this is resulting in one way audio. Our reseller provided ways to override the Shoretel setting (rebooting all devices, Shoregear switch, Shoreware Director, and Shoretel IP 530 phone in that sequence) so it does not use Port 5004 to send RTP out to, but it does not seem to work. Call flow is as follows: Looks like the IP530 phone sends the calls through Shoregear switch to the Broadsoft infrastructure, As the call is answered, the 200 OK comes back from the Broadsoft side; the Edgemarc NATs the 200OK & sends its private IP in the SDP and a Port, however, I see the IP 530 phone stream RTP to the Private IP of the Edgemarc but to port 5004 all the time. Our reseller provided ways to override the Shoretel setting (rebooting all devices, Shoregear switch, Shoreware Director, and Shoretel IP 530 phone in that sequence) so it does not use Port 5004 to send RTP out to, but it does not seem to work. Thx Ujjval Karihaloo

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east coast outage
by beckman＠angryox.com Oct. 20, 2009

Oct. 20, 2009

I'm still getting a lot more busy signals and unable to get through on all of my providers to completely random numbers. I'm unsure if it is related to the Level3 Atlanta issues this morning, since most people are reporting connectivity is back. Anyone else getting reports or experiencing termination strangeness? Also, Scott Berkman, you mentioned that there were reports of L3 issue at NANOG -- you mean at the conference NANOG47, or on the mailing list? Can't find it in the archives: http://mailman.nanog.org/pipermail/nanog/2009-October/thread.html Beckman --------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------

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FW: [outages] Level 3 down in Atlanta
by David_Hiers＠adp.com Oct. 20, 2009

Oct. 20, 2009

David Hiers CCIE (R/S, V), CISSP ADP Dealer Services 2525 SW 1st Ave. Suite 300W Portland, OR 97201 o: 503-205-4467 f: 503-402-3277 From: outages-bounces at outages.org [mailto:outages-bounces at outages.org] On Behalf Of Scott Howard Sent: Tuesday, October 20, 2009 8:20 AM To: outages at outages.org Subject: [outages] Level 3 down in Atlanta Went down about 35 minutes ago. Anyone know what's going on? Scott. This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.

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Shoretel 9.0 Call transfer using SIP softphone Xlite/Eyebeam
by ujjval＠simplesignal.com Oct. 19, 2009

Oct. 19, 2009

I am testing Shoretel shoretel with Broadsoft and Call transfer seems to fail. Eyebeam--> Shoregear--> SIP Provider(Broadsoft)--> PSTN Eyebeam use calls a cell phone on the PSTN side and then tried to transfer the call to PSTN user#2, Has anyone tested this with a SIP Softphone registered to Shoretel ?

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software recommendation
by 28rhills＠gmail.com Oct. 19, 2009

Oct. 19, 2009

Greetings: I'm looking for advice on a project/proof of concept I'm working on. I would like to create a settlement-free peering fabric for voice traffic between and among some peer institutions in my area. Because this is more of a side-project for cost cutting measure, I'm primarily looking at open source software, though commercial product recommendations would be helpful as well. The organizations I would like to connect have their own PBXs with large blocks of numbers (whole NPA-NXXs), with no number portability in or out. I imagine both at the individual PBXs and peering fabric, the number routing would be static. To put another way, we would manually configure which connections the block of telephone numbers is reachable at. Here is the required ASCII art diagram :) +-------+ | PBX | +-------+ | +-------+ | SBC | +-------+ | | +---+ +---+ .--------. +---+ +---+ | P | | S | / \ | S | | P | | B |--| B |------ ( ???? )----------| B |--| B | | X | | C | \ / | C | | X | +---+ +---+ `---------' +---+ +---+ | | +-------+ | SBC | +-------+ | +-------+ | PBX | +-------+ For scalability reasons, a full mesh of connections between and among the SBCs is not an attractive option. Here's what I think I need: Basic SIP routing TCP, TLS, and UDP support What would be nice to have: IPv6 CDR What is probably not needed: User agent client registration, presence, IM, voice mail I see there are several different open source voice projects, ranging from Asterisk to Kamailio, sip-router and others. I'm seeking comments on what you would use for this situation. Are there any existing projects along these lines? Is there one project that is better than another for this application? Thank you

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