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).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:s4XvtaXKNh33jeaNbeFQk35Xw5d1nfForKnSIczODerived trunk in same tenant where the parent trunk is registered
I'd like to find out if it should be possible to use derived trunks in the same tenant in which the parent trunk is configured. My configuration looks like the below: Parent trunk `teamstest.carrier.com` is configured in TAC and connected to a working SBC. The trunk is showing healthy with no issues reported. Base domain and subdomains have been registered in M365, in the same tenant: `teamstest.carrier.com` `account001.teamstest.carrier.com` `account002.teamstest.carrier.com` I am able to create voice routes which use `teamstest.carrier.com` but when I try to create voice routes using the derived trunk, e.g.: New-CsOnlineVoiceRoute -Identity Route1 -OnlinePstnUsages @{add="Account1Record"} -OnlinePstnGatewayList @{add="account001.teamstest.carrier.com"} -NumberPattern ".*" It fails with the error: New-CsOnlineVoiceRoute: Cannot find specified Gateway "account001.teamstest.carrier.com". Is this because I have some incorrect configuration, or is using derived trunks in this manner not supported?
Direct Routing with Local Media Optimization - 6-7 seconds silence
Scenario: Oracle E-SBC (Acme Packet) with internal media interface and public interface for MS Teams Trunk and media. Media Bypass enabled (always) with Local Media Optimisation Client is in corporate network (internal), the media traffic should flow beetween client and internal media interface of the SBC (the Client cannot reach public IP-addresses directly, HTTP only via Proxy, E-SBC media interface is reachable from client network) Problem: Incoming call to MSTeams Client via E-SBC. RTP from caller to callee is fine, but the RTP from the callee to the caller takes 6-7 seconds until it is established. In the SIP flow you can see, that there is a ReINVITE from E-SBC to MSTreams Cloud, directly after the first 200 OK arrives at the SBC. But the Client/MSTeams SBC needs 6-7 seconds to answer the ReINVITE with the 200 OK. After that, everything is fine, and both can hear each other. But it takes 6-7 seconds until the caller can hear the callee. Find attached a schematic SIP sequence diagram with the relevant messages and also a Wireshark capture with absolute times Is that a normal behaviour (it's montioned here, but only for "some cases") or does anybody know how to get around this? Cheers Oliver
Attended call transfer - Enforcing the use of MS Teams infrastructure (vs REFER/Replaces)
In https://learn.microsoft.com/en-us/microsoftteams/troubleshoot/phone-system/direct-routing/issues-with-call-transfers, the following is stated: A call transfer can be made by using any of the following methods, in order of preference: - Using a Session Initiation Protocol (SIP) Refer message. - Using an SIP Invite message that has a Replaces header. This method is mostly used for call queue responses. - Using an internal Microsoft Teams infrastructure. This method isn't visible to SBC. The method is used only if the first two methods are not supported. Assuming I am disabling the first two methods: using REFER request & using "Replaces" header - Disabling REFER by not including it in the "Allow" header - Disabling "Replaces" by rejecting the incoming INVITE from MS SIP Proxy with a 420 response along "Unsupported: replaces" header according to RFC3261 section 8.1.3.5 How can the 3rd method be enforced and let MS Teams infrastructure handles the attended call transfer transparently by bridging the two calls and tearing down the session with MS Teams user who triggered the consultative call? In my setup, I am not including REFER in "Allow" (requests and responses) & I am rejecting the 3rd INVITE received once doing consultative transfer with a 420. However, I end up with a failed call transfer because the 3rd method mentioned in the documentation is not taking place.
