- VoiceOps - lists.voiceops.org

Never seen this on a Polycom
by rdawson＠force3.com April 14, 2015

April 14, 2015

I?ve been dealing with Polycom phones for 8 years and I?ve never seen one with a MAC address that didn?t start with 0004F2, till today . . . an old co-worker of mine sent me a picture of a Soundstation IP 4000 with a MAC address 00E075. I?ve seen thousands of these things deployed, I can?t believe I never ran into this before . . . anyone else seen this? Rob

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Onvoy outage in northern CA?
by tritran＠cox.net April 9, 2015

April 9, 2015

DIDs down for the past hour. -- Tri Tran

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CPEs
by briant＠socket.net April 8, 2015

April 8, 2015

We are currently using the Adtran Netvanta 3100 Series for small SMB hosted pbx locations were we are using someone else's bandwidth (cable, resold dsl, etc), but are looking for something that will give us more insight into the customers connection. These are mainly in situations where we have the customers host/hq but they have remote sites/workers that need to be on the same HPBX. Any boxes out there that you really like? Would like to have centralized management, packet capture, voice quality monitoring.. sub $200 price point.. Thanks in advance, Brian Taylor Socket Telecom, LLC brian at socket.net

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Hiding SIP name on Grandstream display
by caalvarez＠gmail.com April 2, 2015

April 2, 2015

Is anyone here aware of a way to hide the SIP registration name on Grandstream phones? The display shows the account name and actual SIP registration. Some people find that confusing.

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Announcement: Kamailio becomes systemd-rtc-server
by abalashov＠evaristesys.com April 1, 2015

April 1, 2015

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina, summarised the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Device Feature Key Synchronization Exclusive To Broadsoft?
by colton.conor＠gmail.com April 1, 2015

April 1, 2015

Is Device Feature Key Synchronization on Polycom phones an exclusive feature to Broadsoft service providers, or is the same thing implemented on other softswitches such a Asterisk for example? I am talking about if you hit DND on a Polycom phone connected to a broadsoft it will turn on DND on the Broadsoft server. If you go to the Broadsoft severs we interface and turn the feature off, it will instantly show its off on the phone. More info on Device Feature Key Synchronization here http://community.polycom.com/polycom/attachments/polycom/VoIP/2234/1/Device…

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Lighthouse Cablevision
by peeip989＠gmail.com March 27, 2015

March 27, 2015

Anyone in the group from or have a technical contact at Lighthouse Cablevision? Could really use some help in Connecticut. Thanks in advance Pete

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What is your favorite line of IP Phones and Why?
by colton.conor＠gmail.com March 27, 2015

March 27, 2015

What is your favorite line of IP phones and why? What PBX or softswitch do you use these line of phones with? I would love to hear responses for both Hosted PBX applications, and On-Prem PBX applications. Is there any reason or need to use non IP phones in todays environment? I know NEC, Avya, and others make multiple digital non SIP phone systems.

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Landline SMS
by briant＠socket.net March 26, 2015

March 26, 2015

We're looking at offering some SMS/Text Messaging services to our landline customers. I know that this can be done with Broadcloud, but does anyone have any other methods of this that are working for them? Thank you, Brian Taylor Socket Telecom brian at socket.net

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SIP packet capture with index
by nelsonh＠socket.net March 24, 2015

March 24, 2015

I'm looking for options to capture SIP/RTP traffic, index it by call, and make it easy to download the capture for a specific call based on calling/called and time. I want the capture to remain ongoing (rotating capture) with, say, a 96 hour window of calls available. I'm open to hardware and software options. Right now, I have a server that uses tshark running rotating 1-minute captures, but finding and extracting an individual call out of each of the packet segments and merging them together is a slower and more manual process than I'd like, and I'd like to get our techs direct access to these captures as well. Thanks, -- Nelson Hicks Network Operations SOCKET (573) 817-0000 ext. 210 nelsonh at socket.net

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