Do you get any response from Teams at all?

A few base thoughts:


  1.  The first line of your example, it looks like the source port of the 
packet you’re sending is port 5060.  Are you certain that this is sent as TLS?  
I’d normally expect to see an ephemeral TCP port.
  2.  You have outbound calls working, so I’m guessing that you have 
sbc.pride.md as a record when you run Get-CsOnlinePstnGateway, but it would be 
good if you confirmed this.
  3.  Also note that frequently issuing a change to teams returns a positive 
response, but takes a quite a long time to actually become enforced.  For 
example, if you add a new gateway with New-CsOnlinePstnGateway, the commandlet 
may be successful, but it won’t actually work for anywhere from 5 minutes to 
three hours (yes, really, multiple hours…)

We provide service to Teams, but use Ribbon gateways as the ‘last hop’ before 
Teams because they’re “officially” supported, and we’re offering this as a 
commercial service, so customers want to know that they’re using “officially 
supported” solutions.  With that said, we do route the calls to the ribbon 
gateways through teams.


Ben Kaufman
[email protected]<mailto:[email protected]>

From: sr-users <[email protected]> On Behalf Of Bugaian A. 
Vitalie
Sent: Tuesday, February 18, 2020 10:06 AM
To: [email protected]
Subject: [SR-Users] Teams integration

Hello list,

its been about one month for me playing with kamailio and I need some help to 
sort out a real life situation.

I followed this guide https://skalatan.de/en/blog/kamailio-sbc-teams; great 
article, also got some inspiration from here 
https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/..articles look 
very similar.

I manged to get my calls out to kamailio from TEAMS it works perfectly call 
gets connected and  codec negotiation is fine, but I am facing issue geting the 
call inbound to TEAMS.

My invite looks like this:

========================
U 217.26.163.205:5060<http://217.26.163.205:5060> -> 
52.114.75.24:5061<http://52.114.75.24:5061> #22
INVITE 
sip:[email protected]:5061<http://sip:[email protected]:5061>
 SIP/2.0.
Record-Route: <sip:sbc.pride.md:5061;transport=tls;lr>.
Record-Route: <sip:217.26.163.205:5061;nat=yes;lr>.
Via: SIP/2.0/UDP 
217.26.163.205;branch=z9hG4bK7838.7417f50ed201fcada9609f5b7c4e520f.0.
Via: SIP/2.0/UDP 
192.168.169.102:5060;received=46.214.187.67;branch=z9hG4bK80e3f7e5cc50ea11806b6eeeb899592c;rport=5060.
From: "+37379844267" 
<sip:[email protected]<mailto:sip%3a%[email protected]>>;tag=1604785394.
To: "+37360844269" <sip:[email protected]:5061;user=phone>.
Call-ID: 
[email protected]<mailto:[email protected]>.
CSeq: 227 INVITE.
sip:[email protected]:5060;gr=008A94E3-CA50-EA11-805B-6EEEB899592C;alias=46.214.187.67~5060~1Content-Type:
 application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, 
PRACK.
Max-Forwards: 69.
User-Agent: SIPPER for PhonerLite.
Session-Expires: 1800.
Supported: 100rel, replaces, from-change, gruu, timer.
P-Preferred-Identity: 
<sip:[email protected]<mailto:sip%3a%[email protected]>>.
Content-Length:   362.
Contact: <sip:[email protected]:5061;user=phone;transport=tls>.
.
v=0.
o=- 2307737351 1 IN IP4 217.26.163.205.
s=SIPPER for PhonerLite.
c=IN IP4 217.26.163.205.
t=0 0.
m=audio 36864 RTP/SAVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:owGy5+mMZNyO5+9lFmUUOK3WqpEsxJH0+jtWz

===============================================

Anybody has a good invite exaple for Teams?

Or do you see any issue with my invite? I do use :

 
record_route_preset("sbc.pride.md:5061;transport=tls","217.26.163.205:5060;nat=yes");
 add_rr_param(";r2=on");

before sending this out.

Please let me know if you can help.

Thanks.

Vitalie.


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