Anyone here still using Snom phones? Incoming call glitch when using Edge Server
I'm noticing a glitch that is happening with our older Snom 710/720/D725 phones, doesn't affect our Audiocodes phones. I believe it is related to STUN / ICE. When we have an external user that is connecting via our Edge Server (either a remote IP phone or S4B App) and they call an internal extension where the user has a Snom Phone, the audio will drop out after 5-10 seconds. If the internal user with a Snom phone calls that person back, the audio is fine so it is only a one-way issue. Anyone dealt with this issue before?
authenticating with an access token
Has anyone tried authenticating with an access token (using -AadAccessToken or -MsAccessToken)? The old version of New-CsOnlineSession had an -OAuthAccessToken param, which accepted a jwt with the PS permissions in the scope, which one could obtain silently, and so avoid login. New-CsOnlineSession has been dropped and the SfB Connector is end of life in July and Connect-MicrosoftTeams does appear to support token for the telephony functions.
SIP trunk translation issue
Hello, I am using E164 format to call outside, I need to add 99 before full number ( 99+4412345678), I applied the rule on Trunk configuration, however when I make the call I still see +4412345678 without 99 ( I am using Wireshark on mediation server). What I am doing wrong ?
Incoming calls dropping
Hello, Some of the incoming calls are dropping after a few signals if not answered and a few seconds when answered. If answered, both ends of the call can't hear each other. This happens on Polycom VVX phones. User activity report is showing diagnostic ID: 13015. How to fix this? We are using Skype For Business server 2015.
Yealink t55 skype config
Hi All. I hope someone can assist. A client has bought T55's for use with Skype but their calendars are apparently 1 day behind. the phones have had the latest firmware installed and all config seems ok. Calls are flowing without issue. Does anyone have any experience with these devices or can anyone make any suggestions that may have an effect in this instance. NTP has been checked on the device GUI.
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 🙂
Polycom VVX 411 Signing out Users at Random
Hello, We are about to roll out Polycom VVX411 phones using SfB but in our testing users are getting logged out seemingly randomly between 7-30 days. They do get an error "Exchange authentication failed. Sign out and sign in again." on their Polycom phones about 1-2 hours before the phone actually logs them out. We are running Polycom UC Software version 5.9.3.2857 We have MFA enabled for our end users also. We are using SfB and Exchange online, nothing on-prem. Has anyone seen or fixed this issue in their environment? Thanks!Skype for Bussiness - Phone numbers in invitations
Hi, When we create a meeting using Outlook, an email is sending to all participants. But this email only shows our default dial number. Our enterprise has offices in several countries and we want to add all this phone numbers in the emails. How can we did it? We want something like that: Thanks!!Skype for business CU 10 install breaks call transfer on polycom VVX
We have on premise SFB 2 FE 2Edge 2 MediationServer 2 web app 2 SQL always on We use pureip for the gateway and sit web services behind a load balancer All other services use Microsoft recommended DNS load balancing I installed the July CU 10 last week all went well and all DB updates done We rebooted all severs and everything came back online Next days polycom vvx users started to report transfers of incoming PSTN calls not working. We tested the senators and found PSTN calls inbound would reach polycom and could be answered, the use would hit transfer and dial either an internal extension or another PSTN. Immediately phone would say transfer failed. Caller would be put on hold and when users resumes call there would be one way audio. Caller could here user but user could here caller Transfers would work fine from Skype client and mobile app and hold was fine also. ThanksAnalytics / Details on Skype for Business Auto Attendants and Queues
We are using Skype for Business and the Microsoft Phone System Cloud PBX functionality. I'm trying to find some details about some specific calls and not sure where to look. What I have from the PTSN usage report is the details for the ucap_in call to a service number which is an auto attendant. I'm trying to figure out which queue it may have been directed to and if it was answered and by whom and I'm not sure where to look. Can anyone point me in the right direction of where to look for more details?
