18 Feb
2020
18 Feb
'20
7:26 p.m.
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
ben.kaufman@altigen.com<mailto:ben.kaufman@altigen.com>
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Bugaian A.
Vitalie
Sent: Tuesday, February 18, 2020 10:06 AM
To: sr-users(a)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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