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TEAMS Direct Routing Forward - external incoming call forward mobile

Daniel Flieth
Level 1
Level 1

Hello, we have implemented direct routing and noticed that if i do an call forward all in Teams (to direct routing SBC) to my mobile phone and then receive an call from external, the call will not routed to my mobile phone. It seems that this is related to the PAI Format which microsoft is using.

Microsoft is sending the Pai with E164 NUmber and then the userid.

With SIP profiles i was unable to fix this. Does anybody know how to solve this? Maybe my SIP Profiles are just wrong and you have already a solution for this.

This is the incomfing and sent INVITE

*Apr 4 10:14:34.627: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+49XXXXX43276@cube.X.com:5061;user=phone;transport=tls SIP/2.0
FROM: <sip:+49XXXXX24018@sip.pstnhub.microsoft.com:5061;user=phone>;tag=e8440b433eb14911b0ea6fe82310c992
TO: <sip:+49XXXXX43276@cube.XXXX.com:5061;user=phone>
CSEQ: 1 INVITE
CALL-ID: xxx
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK8078dd0a
RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>
CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443;x-i=69e406b5-58a1-4d9d-90f7-259a3589fdc9;x-c=f9b61ea0c0b75e6083b504db5c47b415/d/8/34128e3efb524a32ad0eead331494765>
CONTENT-LENGTH: 1106
MIN-SE: 300
SUPPORTED: histinfo,timer
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2023.3.18.1 i.EUNO.9
CONTENT-TYPE: application/sdp
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
P-ASSERTED-IDENTITY: <tel:+49XXXXX725125>,<sip:u066428@XXXX.com>
PRIVACY: id
SESSION-EXPIRES: 3600

v=0
o=- 22644 0 IN IP4 127.0.0.1
s=session
c=IN IP4 52.113.56.126
b=CT:10000000
t=0 0
m=audio 52144 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118
c=IN IP4 52.113.56.126
a=rtcp:52145
a=ice-ufrag:M6/o
a=ice-pwd:xxx
a=rtcp-mux
a=candidate:1 1 UDP 2130706431 52.113.56.126 52144 typ srflx raddr 10.0.142.225 rport 52144
a=candidate:1 2 UDP 2130705918 52.113.56.126 52145 typ srflx raddr 10.0.142.225 rport 52145
a=candidate:2 1 tcp-act 2121006078 52.113.56.126 49152 typ srflx raddr 10.0.142.225 rport 49152
a=candidate:2 2 tcp-act 2121006078 52.113.56.126 49152 typ srflx raddr 10.0.142.225 rport 49152
a=label:main-audio
a=mid:1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20

*Apr 4 10:14:34.637: //585/550499F480F5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK8078dd0a
From: <sip:+49XXXXX24018@sip.pstnhub.microsoft.com:5061;user=phone>;tag=e8440b433eb14911b0ea6fe82310c992
To: <sip:+49XXXXX43276@cube.XXXX.com:5061;user=phone>
Date: Tue, 04 Apr 2023 10:14:34 GMT
Call-ID: xxx
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-17.2.1r
X-MS-SBC: Cisco UBE/CSR1000/IOS-17.2.1r
Session-ID: 00000000000000000000000000000000;remote=2ecfe6210a8f557480128e7caee5cbaf
Content-Length: 0


*Apr 4 10:14:34.659: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:+49XXXXX43276@10.16.249.111 SIP/2.0
Via: SIP/2.0/TCP 10.4.0.32:5060;branch=z9hG4bK251966
From: <sip:+49XXXXX24018@10.4.0.32>;tag=44664B-F12
To: <sip:+49XXXXX43276@10.16.249.111>
Date: Tue, 04 Apr 2023 10:14:34 GMT
Call-ID: @xxx@10.4.0.32
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1426364916-3523744237-2163588949-4203890751
User-Agent: Cisco-SIPGateway/IOS-17.2.1r
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1680603274
Contact: <sip:+49XXXXX24018@10.4.0.32:5060;transport=tcp>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
P-Asserted-Identity: <sip:+49XXXXX24018@10.4.0.32>
Session-ID: 2ecfe6210a8f557480128e7caee5cbaf;remote=00000000000000000000000000000000
Session-Expires: 3600
Content-Type: application/sdp
Content-Disposition: session;handling=required
do termContent-Length: 257

2 Replies 2

Jonathan Schulenberg
Hall of Fame
Hall of Fame

The PAI header in the outbound SIP INVITE looks fine. What are you saying is wrong with the formatting?

Also, let’s see the SIP profile rule you created for this.

PS- Reminder that SIP profiles are not processed on the incoming call leg by default. You have to enable it under voice service VoIP > sip or the incoming dial-peer.

I agree with @Jonathan Schulenberg that the sent PAI looks okay, but it would of course depend on what format your service provider want to have it in. Do you know how they want to get the various fields, like PAI and so on? If you do and you need to alter any of them I would recommend you to write a SIP profile and test it with the SIP dialog output you have on this page. SIP-Profile Test Tool 



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