pstn calling
29 TopicsCannot resume/end pstn call that has been placed on hold?
Hi All, A weird issue we've had when doing some testing on a tenant. They have direct routing in place with an Audiocodes vSBC and BT SIP trunks. If we place a call in from a mobile to a user's ddi then place the call on hold in the teams client we cannot then resume or end the call. If we do this on an IOS teams client however it works. We seem to only have this issue with the PC client. I have also eliminated the network by trying it on my home environment. Anyone had this? AlexSolved18KViews0likes17CommentsPSTN 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 🙂11KViews0likes3CommentsPolycom 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!9.7KViews0likes15CommentsSIP 2.0 error 503 incoming from cisco CUCM
Hello Team, i have a trunk CUCM to SFB 2015 where skype can call Cisco extension but Cisco cannot call Skype extensions, i get the error 503 service unavailable on the skype server. So far i downloaded and installed the latest SFB update that did not fix the issue. here are some logs of Cisco to SFB fetch from Skype: TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.3D3C::07/01/2019-11:43:13.941.00002000 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [431143657] <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068<-10.10.60.250:35810 INVITE sip:820993@10.10.30.23:5068 SIP/2.0 FROM: "AMBOZOO" <sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979 TO: <sip:820993@10.10.30.23> CSEQ: 101 INVITE CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250 MAX-FORWARDS: 69 VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828 ALLOW-EVENTS: presence CONTACT: <sip:25006@10.10.60.250:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="CSFAMBOZOO";bfcp CONTENT-LENGTH: 202 DATE: Mon, 01 Jul 2019 11:26:00 GMT EXPIRES: 180 SUPPORTED: timer,resource-priority,replaces SUPPORTED: X-cisco-srtp-fallback,X-cisco-original-called USER-AGENT: Cisco-CUCM11.0 CONTENT-TYPE: application/sdp ALLOW: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY P-ASSERTED-IDENTITY: "AMBOZOO" <sip:25006@10.10.60.250> Min-SE: 1800 Session-ID: 00001cc400105000a0001002b529e1f4;remote=00000000000000000000000000000000 Cisco-Guid: 4283652480-0000065536-0000000002-4198238730 Session-Expires: 1800 Remote-Party-ID: "AMBOZOO" <sip:25006@10.10.60.250>;party=calling;screen=yes;privacy=off v=0 o=CiscoSystemsCCM-SIP 40478 1 IN IP4 10.10.60.250 s=SIP Call c=IN IP4 10.123.123.10 t=0 0 m=audio 16386 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------EndOfIncoming SipMessage TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.2FE8::07/01/2019-11:43:13.942.00002001 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [431143657] >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068->10.10.60.250:35810 SIP/2.0 100 Trying FROM: "AMBOZOO"<sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979 TO: <sip:820993@10.10.30.23> CSEQ: 101 INVITE CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250 VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828 CONTENT-LENGTH: 0 ------------EndOfOutgoing SipMessage TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.36B0::07/01/2019-11:43:13.953.00002005 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [431143657] >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068->10.10.60.250:35810 SIP/2.0 503 Service Unavailable FROM: "AMBOZOO"<sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979 TO: <sip:820993@10.10.30.23>;epid=4A4C146B07;tag=707bec76e1 CSEQ: 101 INVITE CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250 VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828 CONTENT-LENGTH: 0 SERVER: RTCC/6.0.0.0 MediationServer ------------EndOfOutgoing SipMessage TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.3D3C::07/01/2019-11:43:14.063.0000200C (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [546807437] <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068<-10.10.60.250:35810 ACK sip:820993@10.10.30.23:5068 SIP/2.0 FROM: "AMBOZOO" <sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979 TO: <sip:820993@10.10.30.23>;epid=4A4C146B07;tag=707bec76e1 CSEQ: 101 ACK CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828 ALLOW-EVENTS: presence CONTENT-LENGTH: 0 DATE: Mon, 01 Jul 2019 11:26:00 GMT USER-AGENT: Cisco-CUCM11.0 ------------EndOfIncoming SipMessage Any help will be appreciated. thank youSolved7.3KViews0likes1CommentPSTN Call Drops if you do consult transfer - Direct Routing
Hi All, I have an issue very similar to the one discussed in this thread: https://techcommunity.microsoft.com/t5/microsoft-teams/pstn-call-drops-if-you-put-the-call-on-hold-then-resume-and-hit/m-p/1626836 started by naimeshmistry We have a very simple setup: Reception AA->Main Reception Queue->Overflow Queue PSTN is done via Direct Routing. 2 x Ribbon SBCs 1000 running FW 8.0.3 trunked to Telstra Australia Around the beginning of March, our receptionist reported that after her Teams client on the reception desktop auto-updated to version 1.4.00.4167 to the next version up the incoming PSTN calls started to drop out during consultative transfer or while putting on hold and resuming off hold. Fortunately, we had a laptop that was stuck with 1.4.00.4167 and she was using that laptop to transfer calls without any issues. Strange enough, we tried to put 1.4.00.4167 back onto her desktop but that did not solve the issue. Since then, we purchased her a completely new desktop and the Teams client is now 1.4.00.7XXX or 1.4.00.8872 have to check. The issue with dropped consult transfer calls is hit and miss. Sometimes she has no issues all day long, sometimes every incoming call drops out when she tries to do consult transfer. Like others mentioned in the original post, we don't have any custom Music on Hold configured in Teams but have one configured on SBCs and this is what gets played back to the person who is on hold. She also has a different headset when she uses the desktop it is a wireless Jabra 900 Pro, when she used her laptop it was Jabra evolve 30 wired one. Looking at the PSTN log on Teams Admin Portal all calls have CallEndReasonLocalUserInitiated Any help or advice would be much appreciated. It drives us and our customers crazy.Solved4.7KViews1like3CommentsInitiate PSTN call from teams to public landline.
Hi all, This is my first post. I search on this concept thoroughly beforehand. But couldn't seem to find anything I have looked into the cloud communications part of the MS #GraphApi in order to make a proof of concept regarding: initiating pstn calls from a 3rd party program through Teams to a public landline. Looking at examples like: https://docs.microsoft.com/en-us/graph/api/call-redirect?view=graph-rest-beta&tabs=http#example-3-forward-a-call-to-a-pstn-number I would imagine that I might be able to something along the lines: User1 in company Contoso activates a method in 3rd party program. Which sends a command to a call bot The call bot initiate a connection to User1 in company Contoso On reply (preferably automatic) The bot calls the designated landline from the 3rd party program When the PSTN call is accepted by UserB in Acme Inc. The call is "merged" and the bot leaves the call. Is this even doable ? Have someone made something similar. If there is another solution that is easier that is also very much welcome. Looking much forward to your comments.3.2KViews0likes10CommentsSkype 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. Thanks2.7KViews0likes7CommentsPSTN callers can't hear us
1. Teams Only organization w/Phone System/Calling Plans 2. Polycom VVX VoIP phones (latest SfB online certified firmware) When someone calls in, the Polycom phones and Teams apps that are logged into that account ring, but when we answer one of the Polycom phones, the person calling us cannot hear us. We can hear them. This happens about 3/4 of the time, but not always, and with seemingly no pattern. It doesn't seem to happen when answering using the actual Teams app - only when using the Polycom phones. I've checked out our firewall settings and we are not blocking any of the documented ports or ranges of ports that need to be open, and there is nothing in the actual call log that indicates something is going wrong. I'm at a loss as to how to even begin troubleshooting this. Any advice as to where I can look for clues? Thanks, Bob2.3KViews0likes2CommentsDial pad not showing on Skype for Business room system
I have newly setup a Lenovo Thinksmart hub for use with Skype for Business. I have assigned a 'Meeting Room' licence and also a 'Domestic and International Calling plan' On system restart, the dialpad option is available until the Skype has signed in then it disappears. I can dial out numbers using the 'Start meeting', but hoped the dialpad would be visible all the time for making normal phone calls. Also the system seems very buggy with the keyboard not wanting to pop up when clicking to add invitees1.7KViews1like0Comments