PSTN calls drop with-in 15 - 30 mins with Audiocodes SBC
Hi Community, Recently, I have come across a scenario worth sharing. PSTN calls drops after 15 mins or 30 mins with Audio Codes SBC or Gateway. This can be related or occur due to numerous reason, however if everything is in place (Network to Configuration) and still the call fails, then you can try the below and if it makes the difference. User makes PSTN call > the flow and connectivity is established > after 15 or 30 mins you see BYE message generated resulting in call drop. changing/modifying the "RTCPActiveCalls" and "EnableSessionTimer" may help or simple does't make any difference. Upon isolation, found that issue is mostly triggered due to a setting in Audio Codes S.B.C "Reset SRTP Upon Re-Key" is enabled (usually disabled by default) By going in to the IP profiles > disabling the setting in AudioCodes the issue was resolved and calls no longer were dropped. P.S - This solution may or may not help you (no harm in trying), But its best recommended to contact support and take assistance if needed 🙂(CDR/QoE) data for Call quality performance - reporting trends and details
I am very interested to understand to hear the <pings> out there - you who use the data from Lync or Skype on-prem's QoE database - whether you consume it in it's raw format in the manner of a data scientist, or consume it through a 3rd-party-developed reporting app in the manner of a customer-experience-support analyst / investigator. Is it just me or does the SfBO reporting still leave something to be desired? There is some elementary reporting, but I am not seeing anything that matched Lync/Skype Monitoring Reports. There is a mountain of meta-data in the QoE database. Is that getting left behind? Microsoft did a good job creating simple report-query-templates in the Monitoring Report server that could render 80% of what most Telecom Managers would expect their team to know from a platform. And for the other 20%, ALL the data was simply... available at the customer's disposal (and/or a partner's disposal). For Voice Engineers, Support Analysts, and Service Owners, this was a dream come true. Performance data at our fingertips. Now, with Skype for Business Online, I seem to bottom out on report category / type templates that I am otherwise accustomed to leveraging in the Monitoring Reports. And I hear partners are unable to get access to the Call Analytics API, so this tells me - simply - that in Skype for Business Online, there is some data I am no longer able to access, that I would like to access. @ Microsoft: Are there still developments happening in this space? I would like to see the SfB Online window unlocked for me to get into my QoE data, so that I (or a partner) can render the data in more relative context, with custom filtering. Some of that data can be used to prove Skype is not the root cause of an issue. Next, I would like to also understand the product team's vision of reporting call quality - for calls made in Microsoft Teams. I guess you have a challenge... because the boil-the-ocean expectation is that we will see all platforms (Skype on Prem + Skype Online + Teams...) to be rendered into a one-report-viewing experience. Right? ;-) What is the road map and pending enhancements for CQD and/or Call Analytics, in general? Where / when does Teams sit on that roadmap? And... Will a customer be able to render their own meta-data or be restricted from it?
Help! My Director is consuming all my resources!
First published on TECHNET on Sep 20, 2017 Author: DJ Ball, Senior Escalation Engineer, Skype for BusinessRecently I worked on a couple of cases where the administrators were reporting higher than average CPU consumption on their Director pool servers.Call queue-Transferring calls to a Sfb user not working
Transferring calls to a Sfb user, when received from a call queue for hybrid users (PSTN on prem), is not working since this week. Anyone experiencing this behavior? "The call couldn't be transferred. Call was no transferred because there was a problem with the calling service" Calling directly SIP user- transferring works. I'm am fully online with PSTN and DID from MS and transferring from the call queue is working for me, this behavior is only for users who have DID and PSTN on prem. I opened a ticket with MSS4B Standard Edition deployement in a branch Site throught VPN Site-2-Site
Hi, I'm currently studying the deployment of Skype For Business. In my case: - I have two sites A and B in two different countries. Let A be the main site. - A Trunk SIP will be deployed only into main site A. - I will have the same Active Directory on both sites and share the same SIP domain. - Currently these two sites are linked together through a Site-2-Site VPN and each site has a firewall (TMG 2010). My objectives: - Have Skype For Business on both sites with full functionality. My constraints: - Use Skype For Business Standard Edition in main site A. - I would like to have a complete installation of skype for business Standard Edition in the branch site B (no SBA and no SBS). My question: 1- Is it possible to bypass the VPN for SIP traffic in the branch B site to avoid SIP performance issues? Thank you!
