Teams integration - sr-users

18 Feb 2020

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 <http://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

ben.kaufman@altigen.com <mailto:ben.kaufman@altigen.com>

*From:* sr-users <sr-users-bounces@lists.kamailio.org
<mailto:sr-users-bounces@lists.kamailio.org>> *On Behalf Of
*Bugaian A. Vitalie
*Sent:* Tuesday, February 18, 2020 10:06 AM
*To:* sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
*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:+37360844269@sip.pstnhub.microsoft.com:5061
<http://sip:+37360844269@sip.pstnhub.microsoft.com: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:+37379844267@sbc.pride.md
<mailto:sip%3A%2B37379844267@sbc.pride.md>>;tag=1604785394.
To: "+37360844269"
<sip:+37360844269@sip.pstnhub.microsoft.com:5061;user=phone>.
Call-ID: 80E3F7E5-CC50-EA11-8069-6EEEB899592C@192.168.169.102
<mailto:80E3F7E5-CC50-EA11-8069-6EEEB899592C@192.168.169.102>.
CSeq: 227 INVITE.
sip:+37379844267@192.168.169.102:5060;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:+37379844267@sbc.pride.md
<mailto:sip%3A%2B37379844267@sbc.pride.md>>.
Content-Length:   362.
Contact:
<sip:+37379844267@sbc.pride.md: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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