August 2012 - VoiceOps - lists.voiceops.org

Maintenance for Performance Technologies SEGway
by scott＠socket.net May 17, 2017

May 17, 2017

We have been using a set of PTI SEGway x201s with point code emulation for a number of years now. We also have an unlicensed spare in stock. The maintenance contract renewals are due and the two quotes I've received are quite high. I know of at least one other company that has chosen to drop the vendor maintenance and take care of that in-house. I would like to hear from anyone that is currently using SEGways and find out what they chose to do for maintenance. I would also be very interested to hear of any bad experiences from chosing the in-house option. Thanks. Scott Stratman Network Facilities Manager SOCKET 573-999-4376 scott at socket.net 1-800-SOCKET-3 www.socket.net

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TCPDump on an Adtran TA908E
by zak＠simplesignal.com Nov. 9, 2012

Nov. 9, 2012

Does anyone know the TCP Dump commands for the Adtran TA908e CLI? *Thanks-* Zak Rupas Tier 3 Engineer Support: 303-242-8606 option 1 Like SimpleSignal on *Facebook <http://www.facebook.com/SimpleSignal> !* SimpleSignal Inc. 3600 S. Yosemite Street Suite 150 Denver, CO 80237 [image: Description: Description: cid:image001.png at 01CBDE2D.5E1CC730]

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Followup: RE: One-way audio on Verizon Business (SIP)
by EWieling＠nyigc.com Aug. 29, 2012

Aug. 29, 2012

Verizon confirmed they are having an issue in their NJ USA rate center. -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Eric Wieling Sent: Wednesday, August 29, 2012 9:48 AM To: voiceops at voiceops.org Subject: [VoiceOps] One-way audio on Verizon Business (SIP) Is anyone experiencing one-way audio on Verizon Business (SIP) service? We have multiple customers experiencing this. The SDP looks fine and we see two-way audio flowing, but one side can't hear anything. We are confident it is not an issue local to us, but it would be great to get confirmation that it is a larger outage. If we re-route to Level 3 the issue does not happen. _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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Hurricane problems?
by David.Hiers＠adp.com Aug. 29, 2012

Aug. 29, 2012

Hi, Is anyone seeing odd problems coming out of Louisiana? We're getting hit with a very unusual call origination/forwarding issue with area code 504. Thanks, David 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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One-way audio on Verizon Business (SIP)
by EWieling＠nyigc.com Aug. 29, 2012

Aug. 29, 2012

Is anyone experiencing one-way audio on Verizon Business (SIP) service? We have multiple customers experiencing this. The SDP looks fine and we see two-way audio flowing, but one side can't hear anything. We are confident it is not an issue local to us, but it would be great to get confirmation that it is a larger outage. If we re-route to Level 3 the issue does not happen.

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Click to dial with Asterisk hosted PBX service
by carlos＠televolve.com Aug. 28, 2012

Aug. 28, 2012

For any of you doing hosted service on Asterisk, do you have a reliable and secure click to dial solution? Particularly for Outlook. -- Carlos Alvarez TelEvolve 602-889-3003

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Bizarre toll-free routing issue
by jay＠west.net Aug. 27, 2012

Aug. 27, 2012

This is driving us nuts. We use Inetwork/Bandwidth/Dash to terminate SIP calls to toll-free numbers. About one out of every 50 calls results in something I have never seen before. The caller hears an IVR of a company other than the one dialed. After a few seconds, it skips to a different, totally unrelated IVR, then another, and so on. It seems to be related to load, occurs more often during peak calling hours, rarely if ever off-hours. We have never gotten connected to a live operator, only IVRs. The jump to a different IVR occurs after the initial greeting if we don't enter any DTMF. The numbers we've had problems with are: 866-841-0143 - AAA 888-792-4900 - Franchise Tax Board (CA) 800-627-8797 - Anthem Blue Cross 877-776-2436 - Drive Insurance I'm told that AAA and Franchise Tax Board terminate on AT&T and Anthem and Drive terminate on Verizon. Getting Inetwork to fix it results in, "An intermediate carrier that shall not be named" (Voldemort?) says that another downstream carrier says that the customer has equipment problems. Examples: Call to 8668410143 supposed to answer "Triple A" went to Pacific Western Sales, jumps to ATT, then Child Support Division of the Office of Attorney General, then Free 411 then Comcast over the 1 minute 51 second duration of the call. 8006278797 duration 1:40, was supposed to be Anthem but answered with Wells Fargo, then skipped to Employment Development Department then Classic Vacations. 8006278797 duration 2:28, was supposed to be Anthem but answered with Wells Fargo, then skipped to Citicard then S.M.U.D. This is tough to troubleshoot. It takes multiple calls to replicate as the vast majority go through without issue. Most of our customers just assume that they mis-dialed and try again. Where we become painfully aware of it is when a customer has one of these numbers on speed-dial and it becomes obvious that it isn't a mis-dial. Has anyone else on the list run across this? Other than "Don't use Inetwork", any ideas on getting it fixed? My fear is that it may not be Inetwork but something beyond them, perhaps the companies involved outsource their call centers to a common hosted service that is having issues.

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SIP Trunk service for 706-778
by abalashov＠evaristesys.com Aug. 21, 2012

Aug. 21, 2012

On 08/20/2012 10:03 PM, Mike Ray, MBA, CNE, CTE wrote: > In my experience, they do act like yokels and do whatever they can to > forestall CLECs. This doesn't surprise me at all. At least in FL, they > will refuse to meet you at the tandem even though they are interconnected > there, and will require local interconnection trunks in the rural town you > want to serve. They know that most of us don't have facilities there, and > this is a very effective barrier to entry. We took it all the way to the > PSC and lost in one rural Florida town. > > So good luck. Unless you've got some sort of facilities in place in this > little town, this is probably gonna hurt. I thought they run their own tandems and expect you to pick up the traffic there? Though, they might have more exacting DEOT requirements for origination than the RBOCs do. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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SIP Trunk service for 706-778
by justlikeef＠gmail.com Aug. 21, 2012

Aug. 21, 2012

All - We have a customer in Northeast GA that we have been trying to get SIP Trunks for almost year now. The problem is that they are in the 706-778- NPA-NXX and the ILEC has successfully prevented anyone from coming in to the area. I have complaints open with various agencies, but even our Representative's office has not been able to get any movement. Is there anyone that can now provide service in the area or could make a suggestion on how to proceed? Thanks, Rob

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VoiceOps Digest, Vol 38, Issue 15
by voiceops＠dbroute.com Aug. 18, 2012

Aug. 18, 2012

Remote provisioning of Grandstream phones is a nightmare. While you can do some of it by a DHCP option, to do everything (phone book, set browser homepage, set timezone, etc,etc), we spent a lot of time writing a perl based system to "watch" for Grandstream phones (by polling the DHCP server looking for the MAC OID), then logging in via HTTP, and basically emulating a human. Would never again do that by choice. Also, we have found they often "go to sleep", it's like the SIP timers are broken, such that when the registration timer expires it **sometimes** doesn't re-register, and if the registrar is restarted it never re-registers. Lastly, when they update firmware, lots of things change, there is little consistency, so we have to re-write Regards....Graeme. http://www.dbroute.com On Fri, 2012-08-17 at 22:34 -0400, voiceops-request at voiceops.org wrote: > 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. Re: Grandstream VoIP phones (Matthew S. Crocker) > 2. Re: Grandstream VoIP phones (Nathan Anderson) > 3. Re: Grandstream VoIP phones (Oren Yehezkely) > 4. Re: Grandstream VoIP phones (Eric Wieling) > 5. Re: Grandstream VoIP phones (Jason Baugher) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 17 Aug 2012 17:06:01 -0400 (EDT) > From: "Matthew S. Crocker" <matthew at corp.crocker.com> > To: "ryandelgrosso at gmail.com" <ryandelgrosso at gmail.com> > Cc: "voiceops at voiceops.org" <voiceops at voiceops.org> > Subject: Re: [VoiceOps] Grandstream VoIP phones > Message-ID: <B0580C6C-992B-490A-87C2-E26B6DCC9A49 at corp.crocker.com> > Content-Type: text/plain; charset=us-ascii > > > On Aug 17, 2012, at 4:31 PM, Ryan Delgrosso <ryandelgrosso at gmail.com> wrote: > > > I agree with polycom on the speakerphone quality front, BUT they are an absolute nightmare in most other aspects. > > > > The time required to reboot a polycom is measured on a calendar, which can be infuriating when your support team is working remotely with a non-savvy customer. > > Not with 4.0.1 firmware > > > > Their web interface is notoriously riddled with security holes (customers do, against our advice place their polycoms bare to the internet and are always surprised when the sip credentials are harvested > > > > All new web interface with 4.0.1 don't know f the wholes have been fixed > > > > The provisioning settings for them are ONLY configurable from the device dialpad, meaning remote configuration is nightmarish. If you have ever listened to a support engineer try to talk someone non-technical through typing out a provisioning URL on that device dialpad you know that it is most certainly one of Dantes levels of hell. > > > > 4.0.1 can be set via the web interface. The have also supported DHCP options for quite a while > > > > I will sacrifice some speakerphone functionality to not place my support engineers in that kind of purgatory, especially compared to devices with reasonably decent management like the Cisco devices. Now if only they could get on board with a solution like Innomedia's DMS product which provides out of band management that punches through NAT, THEN you would have a slam-dunk. > > > > Have you looked at Polycom zero touch provisioning? It is DMS in the cloud > > -Ryan > > > > From a support perspective > > On 08/17/2012 11:26 AM, Scott Berkman wrote: > >> If you care about speakerphone quality and clarity, you should be using > >> Polycom as they destroy everyone else, especially on the models that support > >> HD voice and therefore have the big speaker baffle and better microphones. > >> This is pretty clear when you see that all of Cisco's conference models are > >> just rebranded Polycom's with SCCP firmwares. I'm a big Polycom fan in > >> general, but the Cisco 7940/60 is a close second (just wish it supported > >> more features like subscribe on the 3rd party SIP images). The Cisco's do > >> have some problems with the speakerphone (and handset) at high volume, such > >> as sound still coming out the handset when it shouldn't and clipping. > >> > >> It's been a long while since I touched a Grandstream, but I like using the > >> feel and weight of the handset as an indicator of build quality, and the > >> older Grandstreams failed that test miserably. > >> > >> -Scott > >> > >> -----Original Message----- > >> From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] > >> On Behalf Of Alex Balashov > >> Sent: Friday, August 17, 2012 1:06 PM > >> To: voiceops at voiceops.org > >> Subject: Re: [VoiceOps] Grandstream VoIP phones > >> > >> It's been a while, but my issue with Grandstream has always been: > >> > >> 1) SIP interoperability & stack problems. > >> > >> 2) Low-end speaker phone hardware, thus bad echo cancellation and duplex > >> handling. > >> > >> I mainly judge phone hardware by the quality of its speakerphone, since I'm > >> a very heavy user, so it may be a personal bias. However, by that metric, > >> the Cisco 79xx's and Polycoms win, and Grandstreams, Snoms and Aastras lose. > >> > >> On 08/16/2012 08:07 PM, Carlos Alvarez wrote: > >> > >>> Who else on the list is using them, particularly in a hosted > >>> environment? We've just decided to transition to them as our primary > >>> recommendation instead of the Cisco SPA series. We did it because of > >>> the value and feature set, like having an inexpensive phone with a > >>> small BLF, which a lot of customers asked for. I'm wondering if > >>> others have tips they've learned along the way, or any advice they want to > >> offer. > >>> Also anyone using the advanced features like the browser for > >>> anything useful? > >>> > >>> For those who haven't tried them, or who like us, didn't like their > >>> older models, take another look. We have been surprised at the value > >>> they give us. The prices are low, but the functionality and quality > >>> are high. They aren't Polycom 600s to be sure, but they are nice > >>> phones that have a huge set of features for a great price. Customers > >>> are liking them a lot. > >>> > >>> Has anyone used the new DECT phone? We currently use the Panasonic > >>> DECT phones but they are a nightmare to configure. > >>> > >>> If anyone wants to get in touch with them, our Grandstream contact is > >>> Dennis Ryan, dryan at grandstream.com <mailto:dryan at grandstream.com> . > >>> > >>> -- > >>> Carlos Alvarez > >>> TelEvolve > >>> 602-889-3003 > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> VoiceOps mailing list > >>> VoiceOps at voiceops.org > >>> https://puck.nether.net/mailman/listinfo/voiceops > >>> > >> > >> -- > >> Alex Balashov - Principal > >> Evariste Systems LLC > >> 235 E Ponce de Leon Ave > >> Suite 106 > >> Decatur, GA 30030 > >> Tel: +1-678-954-0670 > >> Fax: +1-404-961-1892 > >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > >> _______________________________________________ > >> VoiceOps mailing list > >> VoiceOps at voiceops.org > >> https://puck.nether.net/mailman/listinfo/voiceops > >> > >> _______________________________________________ > >> VoiceOps mailing list > >> VoiceOps at voiceops.org > >> https://puck.nether.net/mailman/listinfo/voiceops > > > > _______________________________________________ > > VoiceOps mailing list > > VoiceOps at voiceops.org > > https://puck.nether.net/mailman/listinfo/voiceops > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 17 Aug 2012 13:53:14 -0700 > From: Nathan Anderson <nathana at fsr.com> > To: "'voiceops at voiceops.org'" <voiceops at voiceops.org> > Subject: Re: [VoiceOps] Grandstream VoIP phones > Message-ID: > <C12CA40900F4E0448A2D4E66DFA6A59B17D8873A3D at Demekin.FSI.local> > Content-Type: text/plain; charset="us-ascii" > > On Friday, August 17, 2012 1:31 PM, Ryan Delgrosso <> wrote: > > > I agree with polycom on the speakerphone quality front, BUT they are an > > absolute nightmare in most other aspects. > > [snip] > > If you haven't used it yet, a lot of your complaints have been address in UCS 4.0, *especially* the boot-up time, which is MUCH improved. Many configuration options which previously required a reboot of the phone after being changed no longer do. The web management interface got a complete overhaul, too, although I haven't become intimately familiar with the new one since I tend to shy away from using the web management anyway and generally stick to central provisioning. > > In an office environment, you can use DHCP to set the provisioning URL of the phone so that it doesn't have to be entered in manually on the phone itself, although I understand that for hosted PBX service providers, this isn't really an option. What we have ended up doing for telecommuters is to put the phones behind a small, cheap, L2VPN-capable router (MikroTik RB750), have the router tunnel back to the office, and L2-bridge the phone over the VPN so that it can talk to the same DHCP server as all the phones in the office do. Kills many birds with a single stone: phone traffic is encrypted, the office phone switch doesn't need to have a publicly-routable IP assigned to it, the phones themselves aren't behind a NAT from the perspective of the office phone switch, and all phones -- local and remote -- are provisioned the same way. Win, win, win, win. >

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