[SR-Users] Teams integration
Daniel-Constantin Mierla
miconda at gmail.com
Wed Feb 19 08:58:10 CET 2020
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 at 192.168.169.102
> <mailto:800C9970-E650-EA11-808F-6EEEB899592C at 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 at 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 at sbc.pride.md
> <mailto:sbcactivtor at 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 at altigen.com
> <mailto:ben.kaufman at 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 at altigen.com <mailto:ben.kaufman at altigen.com>
>
>
>
> *From:* sr-users <sr-users-bounces at lists.kamailio.org
> <mailto:sr-users-bounces at lists.kamailio.org>> *On Behalf Of
> *Bugaian A. Vitalie
> *Sent:* Tuesday, February 18, 2020 10:06 AM
> *To:* sr-users at lists.kamailio.org <mailto:sr-users at 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 at 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 at sbc.pride.md
> <mailto:sip%3A%2B37379844267 at sbc.pride.md>>;tag=1604785394.
> To: "+37360844269"
> <sip:+37360844269 at sip.pstnhub.microsoft.com:5061;user=phone>.
> Call-ID: 80E3F7E5-CC50-EA11-8069-6EEEB899592C at 192.168.169.102
> <mailto:80E3F7E5-CC50-EA11-8069-6EEEB899592C at 192.168.169.102>.
> CSeq: 227 INVITE.
> sip:+37379844267 at 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 at sbc.pride.md
> <mailto:sip%3A%2B37379844267 at sbc.pride.md>>.
> Content-Length: 362.
> Contact:
> <sip:+37379844267 at 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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--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com
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