Hello guys,
I'm trying to use an SRV as an advertised address but kamailio fails to
start. Is this not possible?
Thanks,
David Villasmil
email: david.villasmil.work(a)gmail.com
Hi everyone. Wanting to see if someone could point me in the right
direction. Still very knew to Kamailio but I am beginning to understand it
better. I'm making an outbound proxy and have everything working well
besides stir/shaken. I'm looking at the module page and have went back and
forth with chatGPT and can't seem to figure this part out. I keep getting
errors on the modparam lines.
Obviously this is a self signed cert because I'm just testing. I am able to
reach and download the cert from the Web server.
Thank you for any assistance.
# SECSIPID for Stir/Shaken
modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key")
modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt")
modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt")
modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2)
route[STIRSHAKEN] {
if (is_method("INVITE")) {
if (!secsipid_add_identity("$fU", "$rU", "A", "", "
http://myIPaddress.com/stir_shaken_cert.crt
<http://myipaddress.com/stir_shaken_cert.crt>",
"/etc/kamailio/secsipid/private.key")) {
xlog("L_ERR", "Failed to sign call with ID: $ci - From: $fU\n");
send_reply("500", "Internal Server Error");
exit;
} else {
xlog("L_INFO", "Successfully signed call with ID: $ci - From:
$fU\n");
}
}
# Relay the call after signing
route(RELAY);
}
Hello All,
I am trying to build a simple proxy that forwards SIP requests to a remote
portaone server. But, kamailio is adding a Via header with received=@IP to
replies got from portaone.
How to disable this behavior as Asterisk (client in my setup) is not
accepting this header.
Thank you.
Hello,
how do i get misctest module in debian packages ? it seems not included in a package.
is compilation the only way to get it ?
im using kamailio 5.7.5 on debian (bullseye) 11.8
version: kamailio 5.7.5 (x86_64/linux)
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled with gcc 10.2.1
Thanks.
Any news on regarding the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available?
Thanks,
Xenofon
________________________________
From: Xenofon Karamanos via sr-dev <sr-dev(a)lists.kamailio.org>
Sent: Wednesday, June 19, 2024 11:51
To: sr-dev(a)lists.kamailio.org
Cc: Xenofon Karamanos <xk(a)gilawa.com>
Subject: [sr-dev] Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org
Hey everybody,
While attempting to use the official Ubuntu package for Kamailio version 5.7.4 on Ubunut 24.04, we encountered an issue where TLS functionality does not seem to work as expected but 5.8 branch does.
Given this, I am inquiring about the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available?
Thank you in advance for your assistance.
Best regards,
Xenofon
I am extremely new at this, but trying to set up TLS with a carrier. TLS connection is good, Invite goes out, we get the 100 and the 200, but subsequent messages (ACK and BYE) are being sent with UDP and I cannot figure out how to get them to maintain the TLS transport. Any suggestions? I think this is the section I'm looking for.
# Manage incoming replies in transaction context
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]") {
route(NATMANAGE);
}
if (has_body("application/sdp")) {
xdbg("rtpengine_manage loop-protect MANAGE_REPLY");
#rtpengine_manage("loop-protect");
}
}
Hi all,
I face a problem when trying to forward an UPDATE request to the IMS which
communicates with UDP transport protocol on one side UDP and SCTP on other
side. When I try to send UPDATE towards the UDP side of IMs I see following
error on kamailio side and request cannot be forwarded
Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: ERROR:
<core> [core/udp_server.c:597]: udp_send(): sendto(sock, buf:
0x7fc180deb5f0, len: 1509, 0, dst: (172.18.0.20:5060), tolen: 16) - err:
Invalid argument (22)
Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: CRITICAL:
<core> [core/udp_server.c:603]: udp_send(): invalid sendtoparameters#012one
possible reason is the server is bound to localhost and#012attempts to send
to the net
The UPDATE message which is tried to be forwarded is as follow:
Here there are double rr one is UDP and one is sctp (Route: <
sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <
sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>)
)
That would be great if you can help why this UPDATE cannot be forwarded.
By the way double rr is on on kamailio config
Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: DEBUG:
<core> [core/forward.c:577]: forward_request(): Sending:#012UPDATE
sip:alice@10.10.141.188;ob;sbc-id=xpllrk47odqh4pb2ofpib6vsym
SIP/2.0#015#012Via: SIP/2.0/UDP
127.0.0.1:50601;branch=z9hG4bKddab.3554948ce3cfe321729448a2b082c2f3.0#015#012Via:
SIP/2.0/UDP 127.0.0.1:7060;branch=z9hG4bKnHN8Q3ya#015#012To: <
sip:alice@ims.wingcon.com>;tag=49IZjY2irfb#015#012From: <
sip:bob@ims.wingcon.com>;tag=hfdqjj2pmui#015#012Call-Id:
3mzzecWt3Cbgvy#015#012CSeq: 1 UPDATE#015#012Max-Forwards:
70#015#012Contact:
<sip:bob@10.10.141.188;ob;sbc-id=o7dpxe7qu2r34zwzio5pis2vqi>#015#012Content-Type:
application/sdp#015#012Route:
<sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>,
<sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <
sip:mo@172.18.0.20;lr;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, <
sip:mo@172.18.0.20;lr;transport=sctp;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, <
sip:mo@172.18.0.120;lr>#015#012Session-Expires:
180;refresher=uas#015#012Supported: replaces, 100rel, timer,
norefersub#015#012Min-SE: 90#015#012P-Charging-Vector:
icid-value=e277783432fd11efba03fa163eed425f;icid-generated-at=172.18.0.120;orig-ioi=172.18.0.120;term-ioi=172.18.0.120#015#012User-Agent:
PJSUA v2.14 Linux-5.4.0.172/x86_64/glibc-2.35#015#012Content-Length:
409#015#012#015#012v=0#015#012o=- 3928313948 3928313950 IN IP4
172.18.0.120#015#012s=pjmedia#015#012b=AS:84#015#012t=0
0#015#012a=X-nat:0#015#012m=audio 30106 RTP/AVP 8 96 97 121 120#015#012c=IN
IP4 172.18.0.120#015#012b=TIAS:64000#015#012a=ssrc:1986855169
cname:1fca52e812ced466#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96
AMR/8000#015#012a=rtpmap:97 AMR-WB/16000#015#012a=rtpmap:121
telephone-event/16000#015#012a=fmtp:121 0-16#015#012a=rtpmap:120
telephone-event/8000#015#012a=fmtp:120
0-16#015#012a=sendrecv#015#012a=rtcp:30107#015#012.
Regards
Serkan
I have a scenario where Kamailio is receiving a retransmit of a 200 OK to a late offer INVITE after it has sent the ACK, and after it has begun to handle a reInvite from the calling party. This results in the following commands to the rtpengine: Offer (reInvite), Offer (200 retransmit), Answer (ACK to retransmitted 200). At this point, I see STUN binding errors both in Chrome, where the webRTC client (called party) is running, and in the rtpengine logs.
ERROR:port.cc(498)]: Received STUN BINDING error response: class=4 number=1 reason='Unauthorized'
[core] STUN authentication mismatch from x.x.x.x:63396
[ice] Received ICE/STUN response code 487 for candidate pair dl2efuRG06eK4nFs:2713745946:1 from x.x.x.x::64509 to x.x.x.x
[ice] ICE role change, now controlled
[ice] Recalculating all ICE pair priorities
[ice] Triggering check for dl2efuRG06eK4nFs:2713745946:1
[ice] ICE/STUN response with unknown transaction received (from x.x.x.x:64509 on interface x.x.x.x:16184)
So, briefly, the call flow up to the point of error looks like this:
INVITE w/o SDP
200 OK w/SDP, Offer to rtpengine
ACK w/SDP, Answer to rtpengine
reInvite w/SDP, Offer to rtpengine
retransmission of 200 OK, Offer to rtpengine
ACK to retransmitted 200, Answer to rtpengine
I have dialog tracking enabled in my config, and it seems like Kamailio is behaving as expected when the retransmitted 200 and ACK are forwarded on, but I'm curious if other people have run into a scenario like this and how they have dealt with it.
Hello,
I need to implement Event: presence, application/pidf+xml as described in
RFC-3856. I understand I need to use the presence_xml module and generate
PUBLISH requests to update the information and ensure that the NOTIFY get
sent. I know nothing abour XCAP or wether I need it or not. Any
information source out there that could help me?
Regards,
Michel Pelletier
Hello,
I noticed that Kamailio can route either to IP and FQDN.
This means the server certificate CN is not checked by the client.
How to enable something like the 'verify_peer' option?
Thanks