Hi,
Is there any other module need to be added for xmlrpc for using
dispatch_rpc().
Because I am getting an below error:
0(22485) ERROR: <core> [core/cfg.y:3274]: yyparse(): cfg. parser: failed
to find command dispatch_rpc (params 0)
0(22485) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in
config file /usr/local/etc/kamailio/kamailio_final.cfg, line 640, column
17: unknown command, missing loadmodule?
I have loaded xmlrpc as well as xhttp modules. Do I need to enable some
other modules as well??
Please Guide me.
Thanks & Kind Regards,
Logeshwaran G
Hello.
I have not been able to find anything in the wiki, or documentation which
covers this.
The only item I have found is related to a module.
>From the information, I have gleaned on this, it is clear an IP address, or
a domain name is to be set here.
What is not clear, is if the server is on multiple IP address whether they
are on a WAN, LAN, or both, if multiple IP addresses are need, or can be
added.
I have a situation where I have a public WAN address, and a LAN address.
Thanks a bunch!
Hi,I am new to kamailio.
when i restart kamailio,I found errors on log.please help through this.
6(5633) ERROR: mymodule [my_mod.c:373]: outbound_call_process():
[outbound_call_process] URI : sip:11111111112@107.170.69.191:1008 received
ip 183.82.115.246 in network ips:107.170.69.191:6090|c1|c2
6(5633) ERROR: mymodule [my_mod.c:379]: outbound_call_process():
[outbound_call_process] Request URI : 11111111112
6(5633) ERROR: mymodule [my_mod.c:395]: outbound_call_process():
outbound_call_process, obcalls privilges :select u.ob_call_privileges,
a.phone_number, replace(a.name,'.','-') from account_info a, users u,
subscriber s where a.id=u.account_id and u.account_id=s.account_id and
u.extension=s.extension and s.username='dyan-dayan' 6(5633) ERROR: mymodule
[my_mod.c:425]: outbound_call_process(): outbound_call_process query from
user:dyan-dayan, phone_type:1 acc_name:dayan 6(5633) ERROR: mymodule
[my_mod.c:501]: outbound_call_process(): [outbound_call_process]NEW URI :
sip:ivr-1-dayan@107.170.69.191:6090
6(5633) ERROR: <core> [udp_server.c:448]: udp_rcv_loop():
recvfrom:Received encrypted packet len:4, v▒..
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SIP and WebRTC*
Scaling your FreeSWITCH platform to serve a growing user base is a critical
challenge. We'll go through the best techniques, practices, and
implementations for Voice and Video Calls, Conferencing, WebRTC, SIP,
Chatting, Presence and Instant Messaging
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Enjoy ClueCon!
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--
Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
Hello,
Can you share the rest of your config (at least, any dmq modparams and the
main route block)?
Cheers,
Charles
On 16 Aug 2017 22:48, "John Petrini" <jpetrini(a)coredial.com> wrote:
Hello,
I've just started doing some testing with dmq but I'm having trouble
getting it to discover all of the nodes in my cluster.
There are 7 kamailio instances with the following in their configs:
Example from the notification server (10.0.10.211):
listen=udp:10.0.10.211:5080
loadmodule "dmq.so"
# ----- dmq params -----
modparam("dmq", "server_address", "sip:10.0.10.211:5080")
modparam("dmq", "notification_address", "sip:10.0.10.211:5080")
Example from one of the nodes (10.0.10.216). (Only the listen and
server_address is changed to match the local ip of each node.)
listen=udp:10.0.10.216:5080
loadmodule "dmq.so"
# ----- dmq params -----
modparam("dmq", "server_address", "sip:10.0.10.216:5080");
modparam("dmq", "notification_address", "sip:10.0.10.211:5080")
Output from kamcmd dmq.list_nodes
{
host: 10.0.10.211
port: 5080
resolved_ip: 10.0.10.211
status: 2
last_notification: 0
local: 0
}
{
host: 10.0.10.216
port: 5080
resolved_ip: 10.0.10.216
status: 2
last_notification: 0
local: 1
}
I'm expecting to see all 7 nodes in the output but only see the local node
and the notification node. I imagine I could use an SRV record for the
notication address to add all of the nodes but I thought the notification
node was supposed to share the rest of the nodes in the cluster.
Can anyone explain what it is I'm doing wrong?
Thank You,
John Petrini
_______________________________________________
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sr-users(a)lists.kamailio.org
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--
Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered
office: Faraday Wharf, Innovation Birmingham Campus, Holt Street,
Birmingham Science Park, Birmingham B7 4BB.
Hello,
I've just started doing some testing with dmq but I'm having trouble
getting it to discover all of the nodes in my cluster.
There are 7 kamailio instances with the following in their configs:
Example from the notification server (10.0.10.211):
listen=udp:10.0.10.211:5080
loadmodule "dmq.so"
# ----- dmq params -----
modparam("dmq", "server_address", "sip:10.0.10.211:5080")
modparam("dmq", "notification_address", "sip:10.0.10.211:5080")
Example from one of the nodes (10.0.10.216). (Only the listen and
server_address is changed to match the local ip of each node.)
listen=udp:10.0.10.216:5080
loadmodule "dmq.so"
# ----- dmq params -----
modparam("dmq", "server_address", "sip:10.0.10.216:5080");
modparam("dmq", "notification_address", "sip:10.0.10.211:5080")
Output from kamcmd dmq.list_nodes
{
host: 10.0.10.211
port: 5080
resolved_ip: 10.0.10.211
status: 2
last_notification: 0
local: 0
}
{
host: 10.0.10.216
port: 5080
resolved_ip: 10.0.10.216
status: 2
last_notification: 0
local: 1
}
I'm expecting to see all 7 nodes in the output but only see the local node
and the notification node. I imagine I could use an SRV record for the
notication address to add all of the nodes but I thought the notification
node was supposed to share the rest of the nodes in the cluster.
Can anyone explain what it is I'm doing wrong?
Thank You,
John Petrini
Hi!
I using uac for send MESSAGE request from kamailio to remote destination
endpoint
Before sending i setting up $fs seudovariable as
$fs = "tls:12.3.4.5:443";
kamailio listeinig this interface and succesfully receives requests and
sending responses on it
When i call uac_req_send();There is error in log file:
ERROR: tm [ut.h:327]: uri2dst2(): no corresponding socket found for
"2.3.26.11" af 2 (udp:2.3.26.11:57050)
ERROR: tm [uac.c:272]: t_uac_prepare(): t_uac: no socket found
If I setting up udp interface for listening then kamailio ognores my $fs
set up and sending message through udp socket.
kamailio version 4.4.6
Now i trying to roll back to 4.4.4 version but not sure if it will be
helpfull.
I'm running Kamailio 4.4.6-2.1 installed from RPM on CentOS 7, where
/var/run is a tmpfs (by default). After every reboot Kamailio fails to
start with the following error:
Aug 16 00:27:39 sbc1 /usr/sbin/kamailio[7135]: ERROR: mi_fifo
[fifo_fnc.c:72]: mi_init_fifo_server(): Can't create FIFO: Permission
denied (mode=432)
I have the following line in my kamailio.cfg
modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")
At this point the /var/run/kamailio directory looks like this
drwx------. 2 root root 80 Aug 16 00:30 .
drwxr-xr-x. 26 root root 880 Aug 16 00:30 ..
srw-------. 1 kamailio kamailio 0 Aug 16 00:30 kamailio_ctl
If I run "chown kamailio /var/run/kamailio" and "systemctl start kamailio"
I am good to go. Am I missing something in my config? It appears Kamailio
is creating the /var/run/kamailio folder as root, writing the kamailio_ctl
file, dropping root privileges, and then trying to write the kamailio_fifo
file.
Thanks,
Ryan
While looking into HOMER I saw that large INVITEs got truncated in the
webinterface, looking at the HEP traffic these (UDP) messages got
fragmented in transport (asterisk/res_hep) and I started to think these
were not getting reassembled correctly. Only to find out the database
schemas have set the msg size set to varchar(1500).
Is there any good reason for not changing this to a bigger default?
Hi everyone,
we have an installation with:
- 1 kamailio instance - with 2 interface, one with public and one with
private IP (for local communication to the db and asterisk ). We use
keepalived service for failover and so, every interface has 2 IPs: the
real one and the “virtual” one for the keepalived service
- some asterisk instances for transcoding and billing - with one
interface with private IP
- 2 RTP (rtpengine) instances - with 2 interfaces, one with public and
one with private IP
A few times happens a loop on the RTPengine and we could see a lot of
lines like this in the log:
“Too many packets in UDP receive queue (more than 50), aborting loop.
Dropped packets possible”
This loop doesn't stop until all the resources are exhausted and it
results in the freeze of the machine.
We don’t understand the reason why these loops are generated.
We have 2 cases:
1) we send an INVITE to the CPE (a Bria on iOS)
The CPE answers with a "100 Trying", a "180 Ringing" and a few seconds
later sends us 3 packets of "200 ok" with SDP like this:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
178.250.x.y;branch=z9hG4bKb842.29952898c30d6d06b75b916618357ccd.2
Via: SIP/2.0/UDP 192.168.x.y:5090;rport=5090;branch=z9hG4bK7fd100fb
Record-Route: <sip:178.250.x.y;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes>
Record-Route: <sip:192.168.x.z;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes>
Contact: <sip:6051018****@158.148.x.y:3180;rinstance=ffdff5e0142dfd2d>
To: <sip:6051018****@192.168.x.z>;tag=29859d3c
From: <sip:051438****@192.168.x.y>;tag=as26b82ac7
Call-ID: 698200851849894c5874186163abe668@192.168.x.y:5090
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE,
PRACK, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria iOS release 3.9.7 stamp 38887.38893
Content-Length: 230
v=0
o=- 1085831961 3 IN IP4 100.81.231.195
s=Cpc session
c=IN IP4 100.81.231.195
t=0 0
m=audio 65454 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
Could the IP of the media part (100.81.231.195 in the CGNAT
100.64.0.0/10) be the cause of the loop?
If that’s the case, how can we prevent the loop?
2) in the second case the CPE is an Avaya
The Avaya send us an INVITE. (Also in this case in the SDP there is a
private IP)
INVITE sip:04818****@178.250.x.y SIP/2.0
From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f
To: <sip:04818****@178.250.x.y>
Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f
CSeq: 1 INVITE
Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-313081f-257bc48-1472e4a5
Privacy: none
Max-Forwards: 70
User-Agent: OfficeServ 7200
Contact: <sip:0432415453@185.55.z.y:5060;transport=udp>
Allow: REGISTER,INVITE,ACK,BYE,REFER,NOTIFY,CANCEL,INFO,OPTIONS,PRACK,SUBSCRIBE,UPDATE
Supported: 100rel
Content-Type: application/sdp
Content-Length: 289
v=0
o=SAMSUNG_SIP_GATEWAY 39304265 0 IN IP4 185.55.x.y
s=SIP_CALL
c=IN IP4 192.168.7.126
t=0 0
m=audio 30012 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
We reply with a “183 Session Progress” and the Avaya send US an “INFO”
packet (dtmf)
we replay with a “200 OK”; and then, after another “INFO” packet we
replay with a “200 OK” with SDP.
After that the Avaya replies with an ACK
2017/08/02 10:17:32.101121 185.55.x.y:5060 -> 178.250.x.y:5060
ACK sip:04818****@178.250.x.y SIP/2.0
From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f
To: <sip:04818****@178.250.x.y>;tag=as783106e4
Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f
CSeq: 1 ACK
Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-3130831-257ffea-84906e4
Max-Forwards: 70
User-Agent: OfficeServ 7200
Route: <sip:178.250.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61>
Route: <sip:192.168.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61>
Contact: <sip:043241****@185.55.x.y:5060;transport=udp>
Content-Length: 0
And after that the chaos…
Has anyone had a problem like this?
Can somebody help us in order to prevent this loop?
In the attached images you can see:
[0] a call, our machines IPs, an ACK from public interface of our SBC
to its private one; and than a lot of ACKs from the keepalived IP to
the private IP of our SBC.
[1] one of these strange ACKs, they are all different in content, but
similar in structure: multiple Record-Route / Via, truncating too long
message;
[2] a "never ending" BYEs.
After modifying the configuration adding some "listen=IP_ADDR:PORT"
(and of course excluding the keepalive IP) we had no loops but is this
enough?
Why Kamailio generated the loops? Is there something else we should
review in our config?
Any advice would help a lot. Thanks.
[0] https://goo.gl/rWh5db
[1] https://goo.gl/MNShq9
[2] https://goo.gl/6mxepG