Hello,
I am considering to release Kamailio v5.4.4 sometime next week, likely
on Monday, February 15, 2020. This is the usual heads up notification to
see if anyone is aware of issues not yet reported to bug tracker and if
yes, do it as soon as possible to give them a chance to be fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
Hello dears
I'm trying to build an IMS testbed based on kamailio and integrate it with the PSTN network, in order to do that I'm looking for AGCF,MGCF open source solution.
Am I able to do the functionality of those nodes based on
Kamailio,Asterisk or any other open source?
thanks and regards
Eyas
I am trying to get a webrtc setup going. Here is what I have
I have asterisk server at 10.123.244.18. The webrtc works internally from the freepbx UCP application As well as the Raspberry phone allocation. This server doesn't have any nat on it all devices are on local / reouted networks.
I have a Proxy server at 10.123.245.30 address. This server is located in AWS and has an elastic IP.
On this server I have ngiinx that will load the raspberry phone up.
What configuration do I need in kamailio and rtpengine to get this working.
If I forward port 8089 in nginx to the /ws side on my asterisk server I can get a call to bridge but with no audio and the call end at 30 seconds when remote. It works internally fine. Likely beccuae the web browser can get to https://10.123.244.18:8089/ws ports fine.
Thanks.
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I did more tests and found that loose_route() does not find socket by
name given in ;sn param:
Feb 12 10:40:59 lab /usr/bin/sip-proxy[16975]: WARNING: rr [loose.c:799]: rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;sn=ext_tls;lr;n1;savp=avp;pm=0)
In config I have defined:
listen=tls:192.x.x.x:8007 name "ext_tls"
-- Juha
Using a domain name in R-R header may not be such a good idea:
Feb 11 12:12:06 lab /usr/bin/sip-proxy[735]: WARNING: rr [loose.c:799] rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;pm=0;savp=avp;lr)
Or is there some means to tell K, which FQDN maps to which IP address,
when there are many such FQDNs?
-- Juha
Hi,
In my failover setting using Dispatcher module, I mark the GW if fails to response. OPTIONS are being sent every 30 sec:
failure_route[DISPATCH_FAILOVER]{
if (t_is_canceled()) exit;
if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) {
ds_mark_dst("ip");
if (ds_next_dst()) {
t_set_fr(0,1000);
t_on_failure("DISPATCH_FAILOVER");
route(RELAY);
exit;
}
}
}
However, Kamailio still recognize it as active gw ( A.A.A.A is down, no response from OPTIONS being sent)
$ sudo sercmd dispatcher.list
{
SET_NO: 1
SET: {
SET_ID: 1
DEST: {
URI: sip:A.A.A.A:5060
FLAGS: AP
PRIORITY: 0
ATTRS:
}
DEST: {
URI: sip:B.B.B.B:5060
FLAGS: AP
PRIORITY: 0
ATTRS:
}
}
}
Any help would be appreciated.
AL
When request is sent from Kamailio to MS Teams SIP Proxy, the top R-R
URI needs to contain FQDN of Kamailio SIP proxy instead of its IP
address. Document
https://skalatan.de/de/blog/kamailio-sbc-teams
suggest to replace record_route(); call with
record_route_preset("SBC-DNS-DOMAIN:5061;transport=tls", "SBC-IP-ADDR:5060");
That works only in a very simple case where the request came in over UDP
or TCP and SIP Proxy has only one listening address, i.e., SBC-IP-ADDR.
One way to solve the problem might be a new r_r function that would take
FQDN of the top R-R URI as argument or introduction of a pv from where
the current record_route() function would take the FQDN if it has been
set.
Any comments or other solutions?
-- Juha
Dear List
Hope this email finds you all well.
I have followed the below tutorial on how to integrate Kamailio with MS
Teams
https://skalatan.de/en/blog/kamailio-sbc-teams
However i have been facing an issue with MS Teams with direct routing where
MS Teams does not send back an ACK after a 200 OK.
The current inbound call flow is the following:
*MS Teams ==> Kamailio==>Asterisk*
Once Asterisk answers with 200 OK we send this 200 OK back to MS Teams
however, MS Teams just doesn't answer back with ACK and call drops.
The below trace shows the INVITE coming from MS Teams and the 200 OK we
send back.
||||||||||||||||||||
====================
tag: rcv
pid: 26135
process: 23
time: 1613025911.983278
date: Thu Feb 11 08:45:11 2021
proto: tls ipv4
srcip: 52.114.132.46
srcport: 4352
dstip: SBC_IP_ADDR
dstport: 5061
~~~~~~~~~~~~~~~~~~~~
INVITE sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0
FROM: Phillip Kyriacou<sip:+357XXXXXXXX@sip.pstnhub.microsoft.com:5061
;user=phone>;tag=c3da9477e05d45fca31c24a155af3318
TO: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone>
CSEQ: 1 INVITE
CALL-ID: 9cb9a2f3144e594c87bccda76791c28e
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45
RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443
;x-i=6e5254d9-79c2-4657-b54b-68791aa8e81f;x-c=9cb9a2f3144e594c87bccda76791c28e/d/10/e21a
6ff7125e4cbd91e75ec687fd4c5d>
CONTENT-LENGTH: 1097
MIN-SE: 300
SUPPORTED: timer
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.2.9.1 i.USEA.7
CONTENT-TYPE: application/sdp
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
SESSION-EXPIRES: 3600
v=0
o=- 170239 0 IN IP4 127.0.0.1
s=session
c=IN IP4 52.113.40.94
b=CT:10000000
t=0 0
m=audio 51414 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118
c=IN IP4 52.113.40.94
a=rtcp:51415
a=ice-ufrag:BwFs
a=ice-pwd:sFyvnfQ9iH101E80ZKjCxi9+
a=rtcp-mux
a=candidate:1 1 UDP 2130706431 52.113.40.94 51414 typ srflx raddr
10.0.137.19 rport 51414
a=candidate:1 2 UDP 2130705918 52.113.40.94 51415 typ srflx raddr
10.0.137.19 rport 51415
a=candidate:2 1 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr
10.0.137.19 rport 49152
a=candidate:2 2 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr
10.0.137.19 rport 49152
a=label:main-audio
a=mid:1
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:Igc2LVM9FH5kQtkRpHw99H5SB7Rd7eKzJy4gdWJg|2^31
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20
||||||||||||||||||||
====================
tag: snd
pid: 26086
process: 6
time: 1613025917.726664
date: Thu Feb 11 08:45:17 2021
proto: tls ipv4
srcip: SBC_IP_ADDR
srcport: 5061
dstip: 52.114.132.46
dstport: 5061
~~~~~~~~~~~~~~~~~~~~
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45
Record-Route:
<sip:SBC-IP-ADDR:5060;ftag=c3da9477e05d45fca31c24a155af3318;lr=on>
Record-Route:
<sip:SBC-FQDN:5061;transport=tls;ftag=c3da9477e05d45fca31c24a155af3318;lr=on>
Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
From: Phillip Kyriacou<sip:+357XXXXXXX@sip.pstnhub.microsoft.com:5061
;user=phone>;tag=c3da9477e05d45fca31c24a155af3318
To: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone>;tag=as659924a4
Call-ID: 9cb9a2f3144e594c87bccda76791c28e
CSeq: 1 INVITE
Server: MediaGW V1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:357XXXXXXXX@ASTERISK_PUBLIC_IP:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 339
v=0
o=root 411266324 411266324 IN IP4 ASTERISK_PUBLIC_IP
s=ASTERISK
c=IN IP4 ASTERISK_PUBLIC_IP
t=0 0
m=audio 14416 RTP/SAVP 0 8 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:9PcaAGjbemJKTERlFTcVwmLRQoDSEQPxX0L1a6RF
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
||||||||||||||||||||
Am I doing anything wrong here that MS Teams wont respond to my 200 OK? The
Kamailio SBC is showing active on MS Teams and Kamailio dispatcher shows
all the MS Team hubs as AP. I also can't see any errors in the log file
My TLS cfg has the following:
======================
*[server:default]method = TLSv1.2+verify_certificate =
yesrequire_certificate = yesprivate_key =
/etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem
<http://sbc.intelligentsupport.eu/privkey.pem>certificate =
/etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem
<http://sbc.intelligentsupport.eu/fullchain.pem>ca_list =
/etc/ssl/certs/ca-certificates.crt*
*[client:default]method = TLSv1.2+verify_certificate =
yesrequire_certificate = yesprivate_key =
/etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem
<http://sbc.intelligentsupport.eu/privkey.pem>certificate =
/etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem
<http://sbc.intelligentsupport.eu/fullchain.pem>ca_list =
/etc/ssl/certs/ca-certificates.crt*
Thanks very much in advance!
Phillip