Hi, having tsilo + PUSH problem.
My probIem is no matter how I ts_store() the data,
ts_append() cannot find any matching uri
ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record
for sip:david@voice.example.com
In route[LOCATION]
{
ts_store("sip:$tU@voice.example.com")
}
in route[PUSHJOIN]
{
ts_append("location", "sip:$tU@voice.example.com")
}
ts.dump looks like this
{
"jsonrpc": "2.0",
"result": {
"Size": 2048,
"R-URIs": {
"R-URI": "sip:david@192.168.122.7:5060;ob",
"Hash": 57095964,
"Transactions": {
"Transaction": {
"Tindex": 15146,
"Tlabel": 805874386
}
}
},
"Stats": {
"RURIs": 1,
"Max-Slots": 1,
"Transactions": 1
}
},
"id": 43582
}
Thanks
Anthony Alba
hi,
i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode.
its great!
But i need format code. Are there some formatters for kamailio.cfg? It
can be for other editors than VScode.
Marek
Hi all,
I need some help on getting to work my scenario:
I have 3 asterisk servers, and every agents (soft phones like zoiper) is
connecting to which server they need by contacting the PublicIP of each
server. (the problem is that I need to reuse the IP address on others
servers) .
I want if is possible to use a proxy like "haproxy" but for sip to redirect
the agents (softphones),
to their coresponding server.
For example:
- james(a)ast.example.com -> must register to asterisk server 1,
- sarah(a)ast2.exampe.com -> must register to asterisk server 2
- andy(a)ast3.example.com-> must register to asterisk server 3
ast.example.com, ast2.example.com, ast3.example.com = will share all
the same ip,
when the request arrives in the proxy it should check the vhost and
redirect to the coresponding server for registering.
Any hints and how to do this scenario?
Hi,
I am integrating PBX with Teams via Kamailio
Call is working but ACK delivery has an issue
This is the ACK Kamailio receives from PBX
I process the ACK with
handle_ruri_alias();
and
record_route_preset(AS per GUIDE LINES)
But this change is not reflected in RELAY
ACK from PBX
ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538 From: "USER" <sip:13300@PBX_IP>;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac To: <sip:1508@KAMAILIO_IP>;tag=a2ce378b106949cf8bac8f978bb573fb Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca CSeq: 23811 ACK Route: <sip:KAMAILIO_IP:7790;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sbc.KAMAILIO_FQDN.com;transport=tls;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAMAILIO_IP:7790 X-Siptrace-Time: 1606135546 516717 X-Siptrace-Method: ACK X-Siptrace-Dir: in
Hello kamailioers,
Version 5.4.2.
I have a R-URI that I need to change TLS -> TCP in all branches.
I do this in request route before t_relay:
// AoR has multiple contacts, all with transport=TLS
route[REQUEST] {
textops.subst_uri("/transport=TLS/transport=TCP/i")
//parallel forking
//why doesn't this change all branches?
t_relay()
}
If I print out the R-URI in the branch route, I see that only the
first R-URI is changed, in the other branches the R-URI transport=TLS
is not changed to transport=TCP.
Any suggestions on how to get this to work across all Contact addresses?
Thanks
Anthony Alba
Hi Folks,
I was wondering if somebody could help me with an issue. I’m a newbie here, just installing Kamailio sip server.
I’ve enabled TLS, and am trying create a SIP Trunk to external SIP Service which is TLS enabled port 5061.
I’ve configured the following in tls.cfg:
[server:default]
method = TLSv1.2+
verify_certificate = yes
require_certificate = yes
private_key = /etc/kamailio/certs/sbc-private.pem
certificate = /etc/kamailio/certs/godaddy.pem
ca_list = /etc/kamailio/certs/calist.pem
In the section above – ca_list = calist.pem contains all the CA’s and Subordinates of the destination server.
Private_key and certificate are of my own server (public godaddy signed)
[client:default]
method = TLSv1.2+
verify_certificate = yes
require_certificate = yes
private_key = /etc/kamailio/certs/sbc-private.pem
certificate = /etc/kamailio/certs/godaddy.pem
ca_list = /etc/kamailio/certs/godaddyca.pem
In the section above the ca_list is godaddy’s ca and subordinate.
In the wireshark I can see that I’m sending out SIP OPTIONS PING (I’m using dispatcher module).
Then the server replies with tls SERVER HELLO which includes it’s certificate
But for some reason we are rejecting it:
Alert (level: fatal, Description: Unknown CA)
How should I set this up to make sure the remote server CA’s are verified?
Thank you,
Hello,
I'm trying to use the function t_uac_send inside a failure_route as
described in https://kamailio.org/docs/modules/5.3.x/modules/tm.html#tm.f.t_uac_send
to send a CANCEL out.
Although the documentation says "it can include From/To tags" i was
not able to get the from tag provided to be used. Another random
generated one is used.
I've tried the following format without success:
t_uac_send("CANCEL", "$ru", "", "", "From: <$fu>;tag=$ft\r\nTo:
$tu\r\nCall-ID: $ci\r\n", "")
t_uac_send("CANCEL", "$ru", "", "", "From: $fu;tag=$ft\r\nTo:
$tu\r\nCall-ID: $ci\r\n", "")
t_uac_send("CANCEL", "$ru", "", "", "From:
bob(a)kamailio.org;tag=2w3e\r\nTo: $tu\r\nCall-ID: $ci\r\n", "")
In the last one which is copied from the documentation that is how the
>From header going out looks like:
From: <bob(a)kamailio.org;tag=2w3e>;tag=3393f0703fb0ccaca74109ff37de39f5-36d71ef0
Any tips on how to get the From tag passed to the function to be used?
Thanks,
Joao Arruda
Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP
So I m using
rtpengine_manage("replace-origin replace-session-connection DTLS=passive OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off ICE=remove RTP")
1-Asterisk(RTP) -----2-Kamailio/RTPEngine ------ 3- Carrier with SRTP
I detect source IPs and to manage this all.
When call is made from 1 to 3 all works and during call 3 send re-Invite to Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of re-invite from 1.
How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from Callee.
Please advise.