Local Survivability with Operator Connect
Wanted to see if anyone out there has stood up a form of Local Survivability for PSTN access in the event someone loses connectivity back to the Teams tenant or an M365 outage? Not sure if it's on the roadmap to have SBA's work in parallel with OC clients? I'd hope you could have the IP Phone register to both OC and the SBA and in the event OC was unreachable the phone would use the SBA to dial out to the PSTN. I understand we can standup different ISP's for carrier diversity but if both went down to a fiber cut, this client needs access to 911 as they are a city with many recreational facilities for the residents that use them. Any thoughts on how we can roll out a Local Survivability option with Operator Connect would be appreciated . Thanks!
Calls from Microsoft Teams App compared to Teams web app
Hello Bit of a tricky one but I believe it maybe a network issue. I have a situation where the client calls a business contact, goes through to an IVR, press's the required selection and after getting put through to the agent all audio drops from the call. Strange aspect is, if you make the call with the Teams app, it works, make it with the Teams web app and it fails (I have tested this by calling from two numbers that go over completely separate carriers). I have captured network logs and can see a continuous stream of UDP so I know the call is not failing, it seems the audio process is being dropped when being routed through the webb app. Has anyone had this before or would know the difference route/process a teams app phone call makes to a teams web app? I believe it is probably a network issue on the B Party side, but as with most calls, proving that is half the work. I am going to do my own research but not come across clear yet, if anyone has anything to add that would assist in this I would be grateful (as well as if I come up with the answer I will put this down as it is an inconvenience for the client).
Delay transferring out from auto attendant
We have been testing Teams with Direct Routing trunks provided by a carrier partner. All inbound and outbound calls direct to/from Teams users connect without issue. We are using the Teams apps on Windows and iOS - no deskphones are involved yet. I have a simple auto attendant set up which greets the caller with the text-to-speech feature, and then transfers to a queue. This queue plays the default music on hold while alerting queue members. The problem I have is that a caller will dial the number associated with the auto attendant, the call will be answered within a couple of seconds, and then immediately the greeting will be read out and the user will be transferred to the call queue. There is then a silent period of between 5-10 seconds before the music on hold for the queue is heard - this varies with each call, at the lower end it's an acceptable delay, but when it gets to 10s of silence it really isn't workable. If I set the routing on the call queue to put the caller back into the auto attendant after a wait period this happens without any silence - the maximum wait period for the queue is reached and the greeting is played as soon as the music on hold ends. I am not able to determine whether the delay is happening within the auto attendant as it just waits after playing the greeting, or whether it's an issue at the start of the call queue. I opened a support ticket and was told that this delay is expected behavior and it is designed that way - the problem is that it's long enough for people to abandon calls as they think there is a problem. The queue is in conference mode, but this isn't an issue with users answering calls from queues so I don't think that is relevant anyway. Is it true that this delay is expected behavior? Is it expected due to a technical limitation or is it really designed that way?
Teams rejects call with error 422
Hi all, I'm having some trouble with calling from a Swyx pbx to Microsoft Teams direct routing. When I place a call from Swyx pbx to Teams, the call get's denied. When I checked the logs I noticed the following: 422 Session Interval Too Small error Teams is rejecting the call because Swyx is sending a 90 sconds field in the Session Expires Header. I searched the web for adjusting the allow Session Expire field in Teams but can't find anything. According to this document: https://datatracker.ietf.org/doc/html/rfc4028 The rule is to have at least a 90 seconds Header. So Swyx does nothing wrong with that. Can anybody help me with this? Or point me in a good direction. Thanks!Customized name display for Resource Account
Good day, I am struggling to customize a name display when a user makes an outbound call on behalf of a Resource Account. Currently our Resource Accounts are built with a display name of 'CompanyABC Purchasing', 'CompanyABC Operations', etc. As the first 15 characters are delivered to the PSTN, we would like to only display the name 'CompanyABC' instead of 'CompanyABC Purc' and 'CompanyABC Oper'. We attempted to modify the display name by adding five spaces after CompanyABC and then continuing with the remainder of the name, but only one space is shown once the update has been saved. We can build a test Caller ID Policy, but I am unsure how to apply it to the Resource Account or the associated Call Queue we are using for our teams. Any suggestions on how this can be completed? Shane
induce delay on ringback tone
using direct routing with teams phone system ... noticed when a teams client dials out to external pstn number it pretty much straightaway sends calling party the ring back tone ... so there could be 2-3 ring back tones on the calling end (approx. 4-5 secs.) before the called party eventually receives the first ringing tone .. is there a way to induce a delay in that initial ring back tone Teams phone system sends to the Teams client ? i just don't want to start giving ringback tone straight away when 180 ringing from itsp is not received for another good 3-4 seconds. thanks.Notification for incoming calls without options to accept/decline call
Hi everyone. I have a very special customer with a special requirement... I've got asked, if it possible to notify people about incoming calls, without options to accept/decline the call. The customer just wants a simple notification about incoming calls, which show the number/person that is calling. Has someone an idea on how this could be achieved? Thanks for every answer ❤️
Teams doesn't reply ACK for 200 OK
Hi, My SBC domain is sbc.portsip.io and resolved to 218.76.62.10, the SBC private IP is 192.168.0.11, and the TLS transport is 5078 for communicating with Teams, and used port 5069 on TCP for communication with backend PBX. When I dialed from teams, and the PBX extension answered the call, SBC send the below 200 OK to teams, voice is good but the teams is not replied ACK to SBC cause the call is hangup after a while, where is wrong? SIP/2.0 200 OK Via: SIP/2.0/TLS 52.114.32.169:5061;branch=z9hG4bKb34e2036 Record-Route: <sip:AAAAAAEAAAAAAscTwKgAEGIwNTdkZTRiNDIzZGZhZmFmNzViMDAwZTE1MjY3OTlh@192.168.0.11:5069;transport=tcp;lr;drr> Record-Route: <sip:7AcAAAIAAAAAAQFqNHIgqWRjYTM1OWQ2NDJjMmM4ODFhOGI0YjcwMmQ4YWEwYmJl@sbc.portsip.io:5078;transport=tls;lr;drr> Record-Route: <sip:sip-du-a-as.pstnhub.microsoft.com:5061;transport=tls;lr> Require: timer Contact: <sip:sbc.portsip.io:5078;transport=tls> To: <sip:+86101@sbc.portsip.io:5078;user=phone>;tag=2f928937 From: "Thomas Oliveri"<sip:+1001@sip.pstnhub.microsoft.com:5061;user=phone>;tag=6ce7fe294b6641e194ea839c798d6295 Call-ID: 677fb489e2ac5661b78a1339affa2029 CSeq: 1 INVITE Session-Expires: 3600;refresher=uac Accept-Language: en Allow: REGISTER, INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE, PUBLISH Content-Type: application/sdp Server: PortSIP SBC v10.0.0.22 Supported: replaces, norefersub, tdialog, join, timer User-Agent: PortSIP UC - Call Manager 16.0.0.175 X-Session-Id: 611040612304556032 X-Trunk-Name: Teams Direct Routing X-CID: 2vaVDqa6gvIvCq5L7Nw_Fw.. Content-Length: 387 v=0 o=- 420019946 1 IN IP4 192.168.0.16 s=ps c=IN IP4 218.76.62.10 t=0 0 m=audio 30006 RTP/SAVP 18 0 8 101 c=IN IP4 218.76.62.10 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=mid:audio a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:s4XvtaXKNh33jeaNbeFQk35Xw5d1nfForKnSIczO
