- VoiceOps - lists.voiceops.org

Cisco 79xx Phones Running SIP and DNS SRV
by zak＠simplesignal.com March 12, 2013

March 12, 2013

Hello- Is anyone out there using Cisco 79xx phones with DNS SRV? I am struggling and due to Cisco not supporting the SIP loads now it?s very hard to find info or insight. Let me know if you?re willing to share please J Zak Rupas Tier 3 Voice Engineer IP Phone/Text Msg: +13032428606 Like SimpleSignal on *Facebook* <http://www.facebook.com/SimpleSignal>* !* SimpleSignal Inc.

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Call quality issue / survey
by nathana＠fsr.com March 11, 2013

March 11, 2013

Hey gang, Let's play a game. I have made two audio rips from two RTP captures I made of two phone calls through two different termination carriers to the same phone number (in this case, it happens to be to the IVR at City Hall of Fairbanks, AK). I have temporarily allocated 2 DIDs for the purpose of this game, and depending on which one you call, you will either hear played back to you the recording of the call as it happened through one carrier, or through the other. Whoever wants to participate, please place calls to both numbers (do it either from a landline or a G.711u VoIP session, please), and tell me which version of the recording you prefer, why you prefer it, and if you had to put it into words (which you do), how you would describe the difference between the two recordings. I am not going to tell you which one I prefer nor which carrier terminated which recording (I'm actually not 100% sure yet in the case of one of them), or even which carriers are involved, at least until after I've gotten some feedback from you all first. Basically, I'm trying to find out if I'm crazy, or if there really is a difference between the two recordings in terms of audio quality. I say there is a stark difference. I've had others tell me they can't hear it. Maybe I have "golden ears". :-P Here are the two phone numbers to call: 208-301-5083 208-301-5084 Alternatively, for those of you would like copies of the actual audio files to listen to, I can provide those upon request; just contact me off-list. Thanks, -- Nathan Anderson First Step Internet, LLC nathana at fsr.com

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Call quality issue / survey
by nick＠flhsi.com March 11, 2013

March 11, 2013

This was my experience as well. 5083 is better. Nick Olsen Network Operations (855) FLSPEED x106 ---------------------------------------- From: "Gabriel Gunderson" <gabe at gundy.org> Sent: Friday, March 08, 2013 9:29 PM To: "Nathan Anderson" <nathana at fsr.com> Cc: "voiceops at voiceops.org" <voiceops at voiceops.org> Subject: Re: [VoiceOps] Call quality issue / survey On Fri, Mar 8, 2013 at 6:55 PM, Nathan Anderson <nathana at fsr.com> wrote: Calls placed from XO eSIP and AT&T cell, exact same results: > 208-301-5083 :00 No ringback :01.5 Music starts :03 Silence and then woman speaking on the IVR. Consistent volume > 208-301-5084 :00 No ringback :07 Loud click :10 Her voice is very weak and fades in (and is somewhat choppy) :12 Normal from that point forward 208-301-5083 is far and away better. Sounds better, starts quicker, no awkward silence. What do I win for playing? Best, Gabe _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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VoIP traffic survey
by frnkblk＠iname.com March 9, 2013

March 9, 2013

I've been asked to address some of the technical matters of the VoIP at a regulatory-oriented meeting, but not being a consultant my knowledge and experience aren't as broad as they should be. One of the concerns by RLECs related to AT&T (and Verizon's) de-regulation initiative is compensation for origination and termination of traffic. Today many non-pricecap rural carriers are paid by IXCs both for any traffic originated or terminated from the network. Most, if not all, of that compensated traffic, is originated as TDM. The concern is that if the rural carriers move to IP that they won't be paid (hardly) anything for origination or termination. Discussions are being held to use ETS (Ethernet Transport Service) and other NECA-type approach to facilitate a way for RLECs to bill for originating and terminating traffic in an IP-world. These private connections, with QoS support, would allow for both capacity and perhaps volume-type billing. But it's been my experience that much of the VoIP long-distance traffic flows over the Internet, not over private Ethernet circuits. My question to this group, which is very IP heavy and therefore much ahead of the curve in terms of IP-oriented telco operations, is how much traffic flows over public (i.e. Internet) circuits versus private (point-to-point to another carrier or VoIP exchange) circuits, and also how that compares inbound versus outbound. http://www.surveymonkey.com/s/6PRCGWM Any additional commentary on this forum would be help. Regards, Frank

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What does an ALG actually do?
by johnl＠taugh.com March 6, 2013

March 6, 2013

I realize that an ALG is a hack in a router that is supposed to allow SIP packets to go through a NAT router. I also realize that for modern SIP equipment, ALG usually causes more problems than it solves, and that it's described in RFCs 2663, 3424, and others. What I can't find anywhere is what a SIP ALG actually does to the packets. Is that written down anywhere, or is it just network folklore?

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Network Security Audit
by dwhite＠olp.net March 6, 2013

March 6, 2013

I have been asked by management to find options to bring in an external auditing company to evaluate our network and provide appropriate recommendations. We are a small telecommunications company looking for firms with experience in Voice (VoIP) security audits, as well as general security experience appropriate for a Broadband ISP, with an internal corporate network. We're located in the southern US (Oklahoma). Public and private recommendations are welcomed. Thank You, -- Dan White

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VoIP traffic survey
by jra＠baylink.com March 4, 2013

March 4, 2013

----- Original Message ----- > From: "Frank Bulk" <frnkblk at iname.com> > My apologies, I'm not following you here -- can you give me an example > of a TDM traffic (i.e. T-1 or DS-3) flowing over the Internet? Didn't say it had to be TDM. I am a VoIP end-user. I pick up my desk phone to make a call, and my PBX routes it. It might go: * To a local PSTN gateway, and out over traditional TDM (or analog) lines, or * To a commercial gateway provider, or * Direct to the other end's PBX as a 'native' VoIP call In either case, it might get there over * Private, dedicated IP networking, or * The Internet. The 6 sextants are "who operates the gateway" multiplied by "who owns the wires I use to get there". Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://baylink.pitas.com 2000 Land Rover DII St Petersburg FL USA #natog +1 727 647 1274

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AT&T Wireless call routing problem / contact
by nathana＠fsr.com March 4, 2013

March 4, 2013

Hey, Does anyone know anyone / have any contact information for ATT Wireless engineering, or (even better) is there an ATT Wireless engineer who would mind contacting me off-list? We're seeing a problem with call routing to certain NXX, but only when forwarding calls to TNs in that NXX. Thanks, -- Nathan Anderson First Step Internet, LLC nathana at fsr.com

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Issue with one touch call list buttons on Polycom 501/Broadsoft
by RDawson＠alliedtelecom.net March 1, 2013

March 1, 2013

We picked up a customer that had a number of 501s. I downloaded the latest firmware, bootrom and configs and created a new device type. Phones boot, provision and register fine however there is no icon displayed for the registered lines and the one touch call list feature does not work. If I boot against the vanilla configs everything is normal, as soon as I boot against the "templated" configs it breaks. I have literally stripped everything out of the configs that I can, they have basic registration information at this point. I have made up dozens of Polycom configurations in the past and never seen this issue. Of course since the device has been out of warranty since 2007 I have to open a PPI ticket so I just wanted to see if anyone may have seen something similar in the past. Thanks, Rob

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You MUST attend SIPNOC. You SHOULD propose a presentation.
by lindsey＠e-c-group.com Feb. 27, 2013

Feb. 27, 2013

You are REQUIRED to attend SIPNOC 2013, April 22-25, 2013 in Herndon, VA. It's a lot like voiceops, but with colocated participants. If an asteroid were to hit SIPNOC, the WebRTC Apocalypse would immediately ensue. In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as described in BCP 14, RFC 2119 and indicate requirement levels for compliant SIPNOC attendees. You MAY send a delegate to represent you at other technical conferences (or user group meetings) scheduled to occur at the same time. Watching a live stream of other concurrent events is OPTIONAL. You SHOULD propose a talk to present at SIPNOC to explain some interesting topic or challenge you've overcome. The talk MUST NOT be a marketing talk (except to brag on your own superpower VoIP skills.) It is RECOMMENDED that the talk include in-depth technical content. Acting in a effortlessly nerdy manner while at SIPNOC is RECOMMENDED; fortunately, 98% of attendees have accomplished this in past years. 2% had to apply effort to fit in. Talking smack about products and vendors or is OPTIONAL. It is RECOMMENDED that some technical representatives from vendors be sent to participate. The following vendors MUST send representatives to appear in person, because their products will surely be discussed: Sonus Snom Sansay Sangoma Polycom Metaswitch Grandstream EdgeWater Digium Cisco BroadSoft Adtran Acme Packet Aastra This email MUST NOT be construed as authorized communication of SIP Forum, the SIPNOC Program Committee, Marc Robins, or any of his cohorts or colleagues. This email does not express the opinions of my clients or my employer (unless they want correct opinions).

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