Hello,
regarding 1), you set request uri (r-uri/$ru) or destination uri ($du)
to an UDP address, but you force sending via a TLS socket. How do you
route out in this case? Do you use dispatcher or some other module?
Cheers,
Daniel
On 18.02.20 19:15, Bugaian A. Vitalie wrote:
Hi Ben,
I get request timeout.
1) Most probably that is the cause> but I just changed it to tls and
get the following>
Feb 18 20:02:05 sbc /usr/local/sbin/kamailio[37664]: WARNING: {1 275
INVITE 800C9970-E650-EA11-808F-6EEEB899592C(a)192.168.169.102
<mailto:800C9970-E650-EA11-808F-6EEEB899592C@192.168.169.102>} <core>
[core/forward.c:229]: get_send_socket2(): protocol/port mismatch
(forced tls:217.26.163.205:5061 <http://217.26.163.205:5061>, to
udp:52.114.75.24:5061 <http://52.114.75.24:5061>)
I forced it to tls on my side as it was udp, but looks like it
complains that other side is udp?
What are tls numbers on msft side? All examples I have show 506, when
I send option it confirms that tls is on 5061 actually>(so above error
is confuzing a bit)
Sent out tm request to TEAMS: OPTIONS
sip:sip.pstnhub.microsoft.com
<http://sip.pstnhub.microsoft.com>;transport=tls SIP/2.0#015#012Via:
SIP/2.0/TLS
217.26.163.205:5061;branch=z9hG4bKcc7f.5a6648b5000000000000000000000000.0#015#012To:
<sip:sip.pstnhub.microsoft.com
<http://sip.pstnhub.microsoft.com>;transport=tls>#015#012From:
<sip:sbc.pride.md
<http://sbc.pride.md>>;tag=2cab7f8b5170c57239647c2c07226d7c-45063c71#015#012CSeq:
10 OPTIONS#015#012Call-ID:
43c771b814737421-37669(a)192.168.172.4#015#012Max-Forwards
<http://43c771b814737421-37669@192.168.172.4#015#012Max-Forwards>:
70#015#012Content-Length: 0#015#012User-Agent: Oracle ESBC#015#012#015#012
2) Its all good and green in web interface and in powershell cli>
PS C:\WINDOWS\system32> Get-CsOnlinePSTNGateway
Identity : sbc.pride.md <http://sbc.pride.md>
InboundTeamsNumberTranslationRules : {}
InboundPstnNumberTranslationRules : {}
OutbundTeamsNumberTranslationRules : {}
OutboundPstnNumberTranslationRules : {}
Fqdn : sbc.pride.md <http://sbc.pride.md>
SipSignalingPort : 5061
FailoverTimeSeconds : 10
ForwardCallHistory : True
ForwardPai : False
SendSipOptions : True
MaxConcurrentSessions : 10
Enabled : True
MediaBypass : False
GatewaySiteId :
GatewaySiteLbrEnabled : False
FailoverResponseCodes : 408,503,504
GenerateRingingWhileLocatingUser : True
PidfLoSupported : False
MediaRelayRoutingLocationOverride :
ProxySbc :
BypassMode : None
Also user is enabled and has correct number in:
PS C:\WINDOWS\system32> Get-CsOnlineUser "sbcactivtor(a)sbc.pride.md
<mailto:sbcactivtor@sbc.pride.md>" | fl *uri*
OnPremLineURI : tel:+37360844269
LineServerURI :
OnPremLineURIManuallySet : True
LineURI : tel:+37360844269
3) Thanks.
Thanks lot for your suggestion. I am looking to get it sorted out at
point 1.
Vitalie.
On Tue, Feb 18, 2020 at 5:29 PM Ben Kaufman <ben.kaufman(a)altigen.com
<mailto:ben.kaufman@altigen.com>> wrote:
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(a)altigen.com <mailto:ben.kaufman@altigen.com>
*From:* sr-users <sr-users-bounces(a)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(a)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(a)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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