18 Feb
2020
18 Feb
'20
9:06 p.m.
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 -> 52.114.75.24:5061 #22
INVITE 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>;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.
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>.
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.
18 Feb
18 Feb
9:26 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.
STATEMENT OF CONFIDENTIALITY:
The information contained in this electronic message and any attachments
to this message are intended for the exclusive use of the addressee(s)
and may contain confidential or privileged information. If you are not
the intended recipient, or the person responsible for delivering the
e-mail to the intended recipient, be advised you have received this
message in error and that any use, dissemination, forwarding, printing,
or copying is strictly prohibited. Please notify AltiGen Communications
immediately at either (888)258-4436 or via email to
administrator(a)altigen.com, and destroy all copies of this message and any
attachments.
11:15 p.m.
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} <core>
[core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced
tls:217.26.163.205:5061, to udp: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;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;transport=tls>#015#012From:
<sip:sbc.pride.md>;tag=2cab7f8b5170c57239647c2c07226d7c-45063c71#015#012CSeq:
10 OPTIONS#015#012Call-ID:
43c771b814737421-37669(a)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
InboundTeamsNumberTranslationRules : {}
InboundPstnNumberTranslationRules : {}
OutbundTeamsNumberTranslationRules : {}
OutboundPstnNumberTranslationRules : {}
Fqdn : 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" | 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> 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 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
*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 -> 52.114.75.24:5061 #22
INVITE 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>;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.
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>.
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.
STATEMENT OF CONFIDENTIALITY:
The information contained in this electronic message and any attachments
to this message are intended for the exclusive use of the addressee(s)
and may contain confidential or privileged information. If you are not
the intended recipient, or the person responsible for delivering the
e-mail to the intended recipient, be advised you have received this
message in error and that any use, dissemination, forwarding, printing,
or copying is strictly prohibited. Please notify AltiGen Communications
immediately at either (888)258-4436 or via email to
administrator(a)altigen.com, and destroy all copies of this message and any
attachments.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
19 Feb
19 Feb
12:58 p.m.
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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_______________________________________________
Kamailio (SER) - Users Mailing List
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
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
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
22 Feb
22 Feb
12:36 a.m.
Hello list,
thanks for support and a good product.
Integration of kamailio and TEAMS work perfectly. He he.
It took me a while to make routing correct, then to find a proper testing
softphone...
PhonerLite works OK only for TEAMS to SIP and Jitsi other way around, but
this is relevant only for lab testing.
P.S. Where to report, bugs for SIREMIS is it another email address?
Vitalie A. Bugaian
24 Feb
24 Feb
5:07 p.m.
Hello,
what was the trick that made it work for you? Anything special that you
can share in terms of configuring either kamailio or teams?
For Siremis I responded in another email.
Cheers,
Daniel
On 21.02.20 20:36, Bugaian A. Vitalie wrote:
Hello list,
thanks for support and a good product.
Integration of kamailio and TEAMS work perfectly. He he.
It took me a while to make routing correct, then to find a proper
testing softphone...
PhonerLite works OK only for TEAMS to SIP and Jitsi other way around,
but this is relevant only for lab testing.
P.S. Where to report, bugs for SIREMIS is it another email address?
Vitalie A. Bugaian
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
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
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
28 Feb
28 Feb
12:04 a.m.
Hi,
the problem was that ngrep or tools I was used to work with, nor sngrep
would show me the invite I am sending out of kamailio because of tls.
So I could not determine if I am sending a proper invite or not, since it
was encrypted...at capture stage.
But, I managed to get siremis with siptrace so I got my invites in mysql
and that helped to fix missing/malformed Record-route headers
and some SDP that PhonerLite was sending, some optional in security, so
Microsoft was rejecting it.(Not acceptable Here)
So I changed user-agent to jitsi and it had proper sdp that msft liked.
Probably a default/samle configuration file with siptrace enabled would
speedup the process, when deploying for first time...
Thanks.
Vitalie A. Bugaian
On Mon, Feb 24, 2020 at 1:07 PM Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
what was the trick that made it work for you? Anything special that you
can share in terms of configuring either kamailio or teams?
For Siremis I responded in another email.
Cheers,
Daniel
On 21.02.20 20:36, Bugaian A. Vitalie wrote:
Hello list,
thanks for support and a good product.
Integration of kamailio and TEAMS work perfectly. He he.
It took me a while to make routing correct, then to find a proper testing
softphone...
PhonerLite works OK only for TEAMS to SIP and Jitsi other way around, but
this is relevant only for lab testing.
P.S. Where to report, bugs for SIREMIS is it another email address?
Vitalie A. Bugaian
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda --
www.linkedin.com/in/miconda
Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
12:53 p.m.
Hello,
thanks for sharing the details!
Check also the sipdump module, can be an alternative to siptrace for tls
traffic troubleshooting, if you don't want to mix up with database storage.
Cheers,
Daniel
On 27.02.20 20:04, Bugaian A. Vitalie wrote:
Hi,
the problem was that ngrep or tools I was used to work with, nor
sngrep would show me the invite I am sending out of kamailio because
of tls.
So I could not determine if I am sending a proper invite or not, since
it was encrypted...at capture stage.
But, I managed to get siremis with siptrace so I got my invites in
mysql and that helped to fix missing/malformed Record-route headers
and some SDP that PhonerLite was sending, some optional in security,
so Microsoft was rejecting it.(Not acceptable Here)
So I changed user-agent to jitsi and it had proper sdp that msft liked.
Probably a default/samle configuration file with siptrace enabled
would speedup the process, when deploying for first time...
Thanks.
Vitalie A. Bugaian
On Mon, Feb 24, 2020 at 1:07 PM Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
what was the trick that made it work for you? Anything special
that you can share in terms of configuring either kamailio or teams?
For Siremis I responded in another email.
Cheers,
Daniel
On 21.02.20 20:36, Bugaian A. Vitalie wrote:
--
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
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
Hello list,
thanks for support and a good product.
Integration of kamailio and TEAMS work perfectly. He he.
It took me a while to make routing correct, then to find a proper
testing softphone...
PhonerLite works OK only for TEAMS to SIP and Jitsi other way
around, but this is relevant only for lab testing.
P.S. Where to report, bugs for SIREMIS is it another email address?
Vitalie A. Bugaian
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com
<http://www.asipto.com>
Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com
<http://www.kamailioworld.com>
1529
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1539
days old
7 comments
3 participants
participants (3)
-
Ben Kaufman -
Bugaian A. Vitalie -
Daniel-Constantin Mierla