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VoiceOps Digest, Vol 22, Issue 5
by Jamie.Montgomery＠comporium.com April 4, 2011

April 4, 2011

Genabnd took over Nortel. They're calling the C2 Jamie Montgomery -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of voiceops-request at voiceops.org Sent: Monday, April 04, 2011 12:00 PM To: voiceops at voiceops.org Subject: VoiceOps Digest, Vol 22, Issue 5 Send VoiceOps mailing list submissions to voiceops at voiceops.org To subscribe or unsubscribe via the World Wide Web, visit https://puck.nether.net/mailman/listinfo/voiceops or, via email, send a message with subject or body 'help' to voiceops-request at voiceops.org You can reach the person managing the list at voiceops-owner at voiceops.org When replying, please edit your Subject line so it is more specific than "Re: Contents of VoiceOps digest..." Today's Topics: 1. Nortel CS 2000 (Hiers, David (DS)) ---------------------------------------------------------------------- Message: 1 Date: Mon, 4 Apr 2011 09:25:34 -0500 From: "Hiers, David (DS)" <David_Hiers at adp.com> To: "voiceops at voiceops.org" <voiceops at voiceops.org> Subject: [VoiceOps] Nortel CS 2000 Message-ID: <81CDFC77FB2DC54DB875D4F8FFC36CE00F2721E196 at DSMAIL2HE.ds.ad.adp.com> Content-Type: text/plain; charset="us-ascii" Hi, What's become of the Nortel CS 2000 under the Avaya flag? Is it a viable platform or is Avaya killing it off? Thanks, 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 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. ------------------------------ _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops End of VoiceOps Digest, Vol 22, Issue 5 ***************************************

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Dialer detection prevention
by Nick.Reifschneider＠360networks.com April 4, 2011

April 4, 2011

Does anyone have information on a good product that can help detect and prevent dialer traffic from hitting a network. We currently limit CPS which helps although that throttles the good with the bad traffic. We don't have a bunch of it however it's enough that we'd like to reduce the volume. Regards, Nick

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"TelePacific Network Outage: Cyber-Terrorism?"
by frnkblk＠iname.com April 1, 2011

April 1, 2011

http://www.channelpartnersonline.com/news/2011/03/telepacific-network-outage -cyber-terrorism.aspx?nck=1 Anyone have more information on this? Didn't seem important enough to make this list, if that's any measure. Frank

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Operator services and Directory assistance recommendation
by mmg＠transtelco.net March 29, 2011

March 29, 2011

Dear list members We are currently looking for a DS/OS service provider that supports SIP. We require the operator services as part of a 911 certification. Could someone recommend a provider? Any advice would be greatly appreciate it Regards

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Truth in Caller ID
by peter＠4isps.com March 23, 2011

March 23, 2011

Did you see this? ERRATUM - RULES AND REGULATIONS IMPLEMENTING THE TRUTH IN CALLER ID ACT OF 2009. Issued an Erratum correcting Notice of Proposed Rulemaking, FCC 11-41, released March 9, 2011. (Dkt No. 11-39 ). Action by: Chief, Wireline Competition Bureau by ERRATUM. WCB http://hraunfoss.fcc.gov/edocs_public/attachmatch/DOC-305331A1.pdf

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SIP Compatible Door Phone
by tim.donahue＠gmail.com March 22, 2011

March 22, 2011

Hi all, We recently got a requirement from a client that needs a door phone that will work with the SIP PBX that we recently installed. I came across the product below for a SIP door phone from Valcom, but I can't find any documentation on Valcom's site. http://superphonestore.com/product_info.php?products_id=7279&cPath=364 Does anyone have good experiences with a door phone that works well with SIP PBX's, preferably with some decent documentation on the web and native SIP vs. using an FXS or FXO and connecting to an analog door phone. Thank you, Tim

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PRI loss/level settings - DTMF issues
by jay＠west.net March 18, 2011

March 18, 2011

On 3/18/11 3:35 PM, Tim Donahue wrote: > Jay, > > We have seen this with the Adtran TA900 series boxes before. Have you > checked that the lbo parameter is set appropriately in the t1 interface > settings? This fixed has fixed echo problems for us in the past. I think we're talking apples and oranges here. LBO controls the level of the digital T1 signal itself. Problems here will show up as CRC or other T-1 errors. The characteristics of the TDM audio encoded in the T-1 in terms of echo, DTMF, etc. are not going to be affected by LBO at all. As long as the T-1 receiver sees the 1.544 Mbits/s stream of ones and zeros, the dB level of the T1 bitstream won't have any effect on level of the u-law DTMF or voice encoded therein. Within the Adtran for FXS lines, and within the Shoretel for the PRI B-channels, there is a configurable level setting for transmit and receive. This is an adjustment in the form of a digital pad or volume control that adjusts the DSP going from the analog to the digital realm and will affect the volume of the signal heard at the earpiece or sent from the mouthpiece. For example, a milliwatt test tone will be measured at the same level at the other end regardless of LBO on the T-1 but will be affected by the configurable gain settings on the FXS or PRI. >From my old Bell-head days, there were standards in the form of a "Loss and level plan" for PBXs connecting to the network to ensure that voice levels are audible and don't result in blasting or "singing". I seem to recall that customer PBXs are recommended for -6 to the network and -3 from the network but can't locate that spec. (This is way different from the -16 and +7 from 0 TLP carrier levels for those of us who have been around for a while...) The Shoretel is currently at +/-0 dB in both directions, and the other party reports that the voice coming from the Shoretel seems noticeably loud. I'm of the opinion to just set it to -6 and -3 and see what happens but was hoping for some verification from the community that this is a reasonable approach. > Other than that, the only reliable solution we have found when these > sort of problems crop up is installing a more configurable box, such as > the Audiocodes Mediant 1000. More expensive, definitely, but we have > replaced 3 or 4 Adtrans that were having problems ranging from echo, to > DTMF not registering properly or duplicating digits, that were deployed > at client sites and the Audiocodes has always been able to fix the > problems they were having. If the main difference is the ability to tweak the levels, we have that in the Shoretel itself. > If you do happen find a solution for controlling the levels on the > Adtran T1 interfaces, I'd love to know what it is. I have contacted > Adtran's support a few times about it and I haven't gotten any > confirmation about if it is possible out of them yet. T-1 signal levels, use LBO. Voice gain from handsets to the T-1 is adjustable on the FXS/FXO interface config. Voice gain to a CAS/PRI, to the best of my knowledge, isn't settable on a TA900. -- Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net Impulse Internet Service - http://www.impulse.net/ Your local telephone and internet company - 805 884-6323 - WB6RDV

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PRI loss/level settings - DTMF issues
by jay＠west.net March 18, 2011

March 18, 2011

We are having some issues with doubled DTMF digits from a customer dialing into conference bridges, etc. Scenario: Shoretel PBX -> PRI -> Adtran TA900 -> SIP trunk. We are using G711/RFC2833. The problem is intermittent, about 5% of calls or less, read-back of received digits shows occasional doubled digits, almost always DTMF "1". We were able to duplicate the issue with the Shoretel system going out a conventional TDM PRI as well, but the customer and the conferencing vendor and the customer tend to blame the problem on "the VoIP system". It is very intermittent and DTMF works fine most of the time. The Shoretel itself sends only a short burst out the PRI regardless of the length of time the DTMF key is held. Comments from voice users on the receiving end are that the Shoretel caller's voice seems "hot". The Adtran has settings to adjust level on FXS but not on PRI. The Shoretel does have adjustments, presently set to 0 dB. It is my recollection from the old Bell/Mitel days that there should be padding on T1/PRI trunks but I can't find a spec. From memory I want to say -6 transmit and -3 receive but Google is failing me on finding this documented. Is there a standard loss/level best practice for this? The fact that the issue occasionally occurs on the TDM trunks leads me to think that the SIP RFC2833 conversion is not to blame and that perhaps there is acoustic coupling between the tone played out the phone's speaker being picked up by the microphone that padding would help. Any suggestions would be appreciated. -- Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net Impulse Internet Service - http://www.impulse.net/ Your local telephone and internet company - 805 884-6323 - WB6RDV

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(no subject)
by mwiles＠akabis.com March 18, 2011

March 18, 2011

Does anyone have experience with VoIP Innovations (www.voipinnovations.com)? Thanks! Mark

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SMS Provider
by abarratt＠socket.net March 17, 2011

March 17, 2011

This is a little off-subject, but thought that because of the underlying SS7 it would qualify. :-D Wondering if anyone out there is purchasing SMS access from a "clearinghouse" such as Sybase or Syniverse/Verisign? Any recommendations? We're looking to develop a paging service for schools, governments, etc. We'd like it to be a Full Long-Code (10 Digit) from our existing number resources... may also have interest in Short-Code (5-6 Digit) SMS for commercial use. Another option would be leasing the services from say Sprint or Verizon as an MVNO -- but don't know if we're ready to go down the road of being a cellular reseller (yet). The Email Gateway is an option, albeit a messy one, as you have to keep track of who's network the end-user's cellphone is on, which could probably be done via an LNP DIP, but we'd rather interface directly to the SMS network. Two way would be pretty bad using this as well. -- Adam E Barratt Sales Engineer SOCKET (573) 817-0000 abarratt at socket.net 1-800-SOCKET-3 www.socket.net ********************************************************************** Privacy Notice: The information in this message is confidential and may be legally privileged. It is intended for use only by the individual or entity to which it is addressed. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you have received this e-mail in error please notify the sender immediately and delete it from your system without copying or forwarding. **********************************************************************

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