Hi,
We have verified if MySQL server is offline when Kamailio is starting, Kamailio fails to start correctly.Is this a bug?
Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [db.c:320]: could not add connection to the poolJun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: usrloc [dlist.c:518]: failed to open database connectionJun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: registrar [reg_mod.c:472]: failed to register domainJun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:858Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:861Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:863
We are running kamailio 3.3.7.
Thanks,Al
Hello,
This is question on PBX behavior, what is the right thing to do, and how do
PBX's generally behave.
If a user on a phone, dials a number, which happens to be configured on the
same phone system (for example another tenant), there are two options:
1. The PBX notices this, and directly connects the phone to the DID on
that system
2. The PBX sends the call out on the SIP trunk, and the provider-routing
sends the call back as an incoming call.
What are the pros and cons of each option? How do PBX's generally behave?
Thanks,
Antonio
Hi All
I am using Kamailio 4.0.4 (x86_64/linux).
I wanted to do simultaneous registrations of large number of endpoints.
Many times I see only 2048 endpoints registered and for other endpoints
Kamailio did not respond for REGISTER message.
I am tcp as transport.
Now my questions are
1. What are the different configurations available to increase Kamailio
performance
2. If I put Kamailio in a Linux 64 Bit, 2GHz, 8GB RAM machine, how much
simultaneous and total registrations can be achieved.
I have tuned system file FDs, tcp related configurations like ephemeral
port range, and other required configurations.
I hope there should be some configurations in Kamailio to achieve higher
throughput , can somebody help me out.
Thanks
Austin
Hi, I noticed thath command core.tcp_list in kamcmd gives output that
does not make sense to me:
---
kamcmd> core.tcp_list
{
id: 1
type: TLS
state: CONN_OK
timeout: 3599
ref_count: 1
src_ip: 192.168.47.132
src_port: 5061
dst_ip: 10.10.235.9
dst_port: 54769
}
{
id: 2
type: TLS
state: CONN_OK
timeout: 3599
ref_count: 1
src_ip: 192.168.47.132
src_port: 32854
dst_ip: 10.10.235.9
dst_port: 5041
}
---
In my scenario I am connecting via TLS to host 192.168.47.132:5061.
And host 192.168.47.132 is connecting to my proxy host 10.10.235.9:5041.
pcap verifies this. Also tls.list gives similar output.
Values regarding incoming connection (id=2) makes total sense.
It seems that values regarding outgoing connection (id=1) have
dst_* values swapped with src_* values.
To debug, I placed this logging to tcp_main.c:
---
$ git diff tcp_main.c
diff --git a/tcp_main.c b/tcp_main.c
index 5830c8e..3717cf6 100644
--- a/tcp_main.c
+++ b/tcp_main.c
@@ -1207,6 +1207,7 @@ again:
}
from=&my_name; /* update from with the real "from" address */
su2ip_addr(&ip, &my_name);
+ LM_WARN("after getsockname() my_name=%s server=%s ip=%s\n",
su2a(&my_name, sizeof(my_name)), su2a(server, sizeof(*server)),
ip_addr2a(&ip));
find_socket:
#ifdef USE_TLS
if (unlikely(type==PROTO_TLS))
@@ -1223,6 +1224,7 @@ find_socket:
else *res_si=sendipv6_tcp;
}
*res_local_addr=*from;
+ LM_WARN("before return my_name=%s server=%s ip=%s\n",
su2a(&my_name, sizeof(my_name)), su2a(server, sizeof(*server)),
ip_addr2a(&ip));
return s;
error:
if (s!=-1) tcp_safe_close(s);
---
This gives following output:
WARNING: <core> [tcp_main.c:1210]: tcp_do_connect(): after getsockname() my_name=192.168.47.132:5061 server=192.168.47.132:5061 ip=10.10.235.9
WARNING: <core> [tcp_main.c:1227]: tcp_do_connect(): before return my_name=192.168.47.132:5061 server=192.168.47.132:5061 ip=10.10.235.9
Are tcp endpoint values really wrong way around or did I undertand
something wrong about their meaning?
I am running Kamailio 4.2.5 (x86_64/linux) 61d84c-dirty from git.
--
Mikko Lehto
Hello,
I'm trying to change contact header in SIP response message by stripping
first characters (for testing only) as below :
$ct = $(ct{s.strip,25});
But I got below error saying that $ct is read-only:
0(15410) : <core> [cfg.y:3419]: yyerror_at(): parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 575, column 5-7: read only pvar
in assignment left side
ERROR: bad config file (1 errors)
The question is how can I change contact header before it's sending of
Kamailio?
Thanks
Ali
I have a questions in regards to SIP Auth, and specifically how Asterisk 13
with PJSIP appears to send Invites.
It seems that the typical Kamailio config sends the SIP auth challenge with
the from domain as the auth realm.
however in the case of the Asterisk 13 ( PJSIP ) invite, the from domain is
an RFC1918 IP address and I would have to use the domain from the request.
What im wanting to know is what the consensus is on where the auth realm
comes from.
does everyone 100% agree that you send auth using the SIP From header, or
is this something that some do Request domain ? What im considering doing
is to use the from domain, but if from domain = RFC1918 then use request
domain.
Im just after everyone's opinion on how to best handle this invite, from a
auth_challenge perspective.
( not so much if the Asterisk 13 PJSIP module is doing the right or wrong
thing )
INVITE sip:555111222333@sip.mytestdomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP
180.111.222.333:5060;rport;branch=z9hG4bKPjafb111a9-aaaf-4c9e-b456-748514855a06
From: <sip:555111222333@172.18.90.1
>;tag=88317259-052b-4997-bd2e-9c8836e27b47
To: <sip:555111222333@sip.netsip.net.au>
Contact: <sip:9d43c8df-6378-4266-89c1-0a0bb0e1d0d1@180.111.222.333:5060>
Call-ID: b7da52e2-dfd1-434e-a2c2-b28e1683ce0e
--
Sincerely
Jay
Hi,
I have added the ASTERISK integration in kamailio.cfg, after adding those,
am not able to start kamailio service, its getting failed, attached is the
script.
Any help would be appreciated.
Thank you with regards,
Gopalkrishnan N.
Mob: +91 99404 91346
VoIP call - sip:saigop@gtalk2voip.com