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Broadsoft to Netsapiens
by colton.conor＠gmail.com Oct. 21, 2015

Oct. 21, 2015

Wondering if anyone has made the transition from a Broadsoft platform to a Netsapiens platform? If so, was wondering what Broadsoft has if anything that Netsapiens does not? Overall I am very impressed with the Netsapiens platform, but I am unsure what is missing in their platform. Is there anything that Netsapiens does not do well that Broadsoft does or vice versa?

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Disconnected numbers and SIP
by abalashov＠evaristesys.com Oct. 20, 2015

Oct. 20, 2015

?Does ISUP have a code for this? Is it consistently signalled in the intermachine world?? What about Q.931? (Sorry for unintentional off-list replies. This list doesn't send a list Reply-To seemingly.) -- 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/ Sent?from?my?BlackBerry.

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Edgewater 4750 loses config after reboot
by tritran＠cox.net Oct. 20, 2015

Oct. 20, 2015

Has anyone had this happened? I'm also seeing certain units losing access to the management interface. -- Tri Tran

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Polycom provisioning for a hosted PBX environment
by caalvarez＠gmail.com Oct. 16, 2015

Oct. 16, 2015

We don't sell/recommend Polycom, but we have quite a few customers coming to us from other VoIP carriers and they already have them. We have built a provisioning system that will use HTTP, and it works just fine, except we're not sure what the best way to secure it might be. The phones can do username/password, but then that means someone has to go put that into every single phone. From what we're able to find, there's no way for option 66 to assign this info permanently (meaning you can remove the option 66 settings after initial config). We don't think we can permanently leave option 66 in place because many customers have a mix of phones, and will need different settings for each type. Also we don't control their internal network, and forcing them to make a permanent change could be challenging. I'd love to hear how some of you in an ITSP/hosted environment are handling this.

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VoIPMonitor with Polycom Phones
by colton.conor＠gmail.com Oct. 14, 2015

Oct. 14, 2015

I know many on the list use the VoIP Monitor application, and we are about to evaluate this application. Most of our send users use Polycom Phones, and have the Polycom Productivity Suite Application: Voice Quality Monitoring (VQmon) license on the phones. According to this Polycom page, With Polycom Productivity Suite Application: Voice Quality Monitoring, IT Managers can monitor and troubleshoot voice quality issues more quickly. - Transmit metrics in real time, or near real time, directly from the phone in the standard *RTCP-XR (IETF RFC 3611) format* - Publish metrics using the SIP PUBLISH method - Provide flexible reporting and alarming based on metric triggers and/or events - Enable quick, real-time, system-wide troubleshooting while the affected customer is still on the phone http://www.polycom.com/products-services/voice/applications/voice-quality-m… My question is does VoIP Monitor collect and support the *RTCP-XR (IETF RFC 3611) format? *I can not seem to find any information claiming it does. I found another page saying the Polycom VQMon software also supports SIP RTCP Summary Reports (RFC 6035). Does VoIPMointor support those?

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Telecom fraud survey
by sahin＠eurecom.fr Oct. 13, 2015

Oct. 13, 2015

Hello, As part of our academic work, we are trying to systematize knowledge on telecom fraud, focusing on voice telephony. For this purpose, we created a high level picture that summarize and organize the fraud ecosystem in different layers, ranging from the inherent flaws in the roots of the system to the final goals of the fraudster. We are now looking for feedback from the industry experts of the field through a survey (should take 5-10 minutes to complete): https://www.surveymonkey.com/r/3CJ7HY9 The responses to this survey will be anonymous and only be used aggregated for research purposes. We will publish a research paper using the results of this study, making this work helpful for both academia and industry. Thank you for supporting our work with your valuable insights. Best regards, Merve

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SIP return message on misdials - 487 or 500/503 or ?
by erickbee＠gmail.com Oct. 12, 2015

Oct. 12, 2015

Hello, Just trying to get idea of what is normal on what providers should return for a misdial, bad unknown number, etc. On one provider, I get a CANCEL and a 487 Request Terminated on mis-dials. On another provider, I get a 503 Service Unavailable and a 500 Internal Service Error back. Is there a proper code to expect or is this up to the provider? I was working on chasing down another issue and noticed a lot of 500 and 503's on calls and they were all misdials (bad numbers, etc) on this one provider. I can't call these numbers from other systems or my cell phone so know they are not good numbers people are calling. Thanks, Erick

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NMS recommendations
by shripald＠gmail.com Oct. 12, 2015

Oct. 12, 2015

Hey All, i'm looking for recommendations for new nms platforms. We currently use one open source platform for CPE and one platform for Core and Infrastructure. I'm looking for a commercial or open source that can handle all devices from Network, Security (Firewalls), Servers (Linux, Windows), ESX, and standard SNMP devices. I need it to be relatively easy to implement and geographically redundant. If i can collapse both CPE and Core into one platform, that would be a bonus. If there is something i should stay away from, please mention that as well. thanks! Shri

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Do you use Bandwidth.com for Termination?
by beckman＠angryox.com Oct. 9, 2015

Oct. 9, 2015

If so, please respond privately. I've got a question about recent SIP responses for you, and if you see what I'm seeing, and also feel that it needs to be resolved, to ask you to open a ticket with Bandwidth to get some movement on the issue. Beckman --------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------

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Cisco 7941 SIP
by peeip989＠gmail.com Oct. 8, 2015

Oct. 8, 2015

Greetings Voice Operators, We have an interesting (code word for annoying) challenge that we've never dealt with before, probably because we don't do much with Cisco phones. We have a new customer coming on who wants to keep their very old Cisco 7941 phones. They have a few offices and the phones work as expected behind an Edgemarc. However, they also have 100+ home users, and that's where the issue comes in. Apparently Cisco introduced a security "feature" where they create the session using a random high numbered port (e.g. 49123) but in the Via header, they say to respond to *private IP, port 5060*. So when the SBC sees the private address it assumes it is being NAT'd through a firewall and replies back to *public IP, port 49123*. What we're seeing is that the home router passes the response back to *private IP, port 49123*, which the phone doesn't accept (because it wants it on 5060) and the REGISTER fails. As you know most home routers are poor at handling ALG (and we've tested and found they are equally bad at handling this scenario). We (and the customer) don't want to troubleshoot 100+ individual home routers. We haven't found a way to turn off this really awesome "feature" so we're trying to find other solutions. Anyone been through this and have any suggestions? Thanks, Pete

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